AlexxIT/go2rtc AlexxIT/go2rtc

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AlexxIT/go2rtc
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AlexxIT

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Repository Details

Ultimate camera streaming application with support RTSP, RTMP, HTTP-FLV, WebRTC, MSE, HLS, MP4, MJPEG, HomeKit, FFmpeg, etc.

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:

  • go2rtc_win64.zip - Windows 64-bit
  • go2rtc_win32.zip - Windows 32-bit
  • go2rtc_win_arm64.zip - Windows ARM 64-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_armv6 - Linux ARMv6 (for old Raspberry 1 and Zero)
  • go2rtc_linux_mipsel - Linux MIPS (ex. Xiaomi Gateway 3, Wyze cameras)
  • go2rtc_mac_amd64.zip - Mac Intel 64-bit
  • go2rtc_mac_arm64.zip - Mac ARM 64-bit

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

go2rtc: Docker

Container alexxit/go2rtc with support amd64, 386, arm64, arm. This container is the same as Home Assistant Add-on, but can be used separately from Home Assistant. Container has preinstalled 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:

  • Binary: GitHub > Actions > Build and Push > latest run > Artifacts section (you should be logged in to GitHub)
  • Docker: alexxit/go2rtc:master or alexxit/go2rtc:master-hardware versions
  • Hass Add-on: go2rtc master or go2rtc master hardware versions

Configuration

  • by default go2rtc will search go2rtc.yaml in the current work dirrectory
  • api server will start on default 1984 port (TCP)
  • rtsp server will start on default 8554 port (TCP)
  • webrtc will use port 8555 (TCP/UDP) for connections
  • ffmpeg will use default transcoding options

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

Available modules:

  • streams
  • api - HTTP API (important for WebRTC support)
  • rtsp - RTSP Server (important for FFmpeg support)
  • webrtc - WebRTC Server
  • mp4 - MSE, MP4 stream and MP4 shapshot Server
  • hls - HLS TS or fMP4 stream Server
  • mjpeg - MJPEG Server
  • ffmpeg - FFmpeg integration
  • ngrok - Ngrok integration (external access for private network)
  • hass - Home Assistant integration
  • log - logs config

Module: Streams

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

Available source types:

  • rtsp - RTSP and RTSPS cameras with two way audio support
  • rtmp - RTMP streams
  • http - HTTP-FLV, MPEG-TS, JPEG (snapshots), MJPEG streams
  • onvif - get camera RTSP link and snapshot link using ONVIF protocol
  • ffmpeg - FFmpeg integration (HLS, files and many others)
  • ffmpeg:device - local USB Camera or Webcam
  • exec - get media from external app output
  • echo - get stream link from bash or python
  • homekit - streaming from HomeKit Camera
  • bubble - streaming from ESeeCloud/dvr163 NVR
  • dvrip - streaming from DVR-IP NVR
  • tapo - TP-Link Tapo cameras with two way audio support
  • ivideon - public cameras from Ivideon service
  • hass - Home Assistant integration
  • isapi - two way audio for Hikvision (ISAPI) cameras
  • roborock - Roborock vacuums with cameras
  • webrtc - WebRTC/WHEP sources
  • webtorrent - WebTorrent source from another go2rtc

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:[email protected]/av_stream/ch0
  dahua_camera:
    - rtsp://admin:[email protected]/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
    - rtsp://admin:[email protected]/cam/realmonitor?channel=1&subtype=1
  amcrest_doorbell:
    - rtsp://username:[email protected]:554/cam/realmonitor?channel=1&subtype=0#backchannel=0
  unifi_camera: rtspx://192.168.1.123:7441/fD6ouM72bWoFijxK
  glichy_camera: ffmpeg:rstp://username:[email protected]/live/ch00_1

Recommendations

  • Amcrest Doorbell users may want to disable two way audio, because with an active stream you won't have a call button working. You need to add #backchannel=0 to the end of your RTSP link in YAML config file
  • Dahua Doorbell users may want to change backchannel audio codec
  • Reolink users may want NOT to use RTSP protocol at all, some camera models have a very awful unusable stream implementation
  • Ubiquiti UniFi users may want to disable HTTPS verification. Use rtspx:// prefix instead of rtsps://. And don't use ?enableSrtp suffix
  • TP-Link Tapo users may skip login and password, because go2rtc support login without them
  • 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
  • If the stream from your camera is glitchy, try using ffmpeg source. It will not add CPU load if you won't use transcoding
  • If the stream from your camera is very glitchy, try to use transcoding with ffmpeg source

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:[email protected]:4567/rtsp-over-websocket
  # WebSocket without authorization, RTSP - with
  dahua-rtsp-ws: rtsp://user:[email protected]/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 Frigate.

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

Source: HTTP

Support Content-Type:

  • HTTP-FLV (video/x-flv) - same as RTMP, but over HTTP
  • HTTP-JPEG (image/jpeg) - camera snapshot link, can be converted by go2rtc to MJPEG stream
  • HTTP-MJPEG (multipart/x) - simple MJPEG stream over HTTP
  • MPEG-TS (video/mpeg) - legacy streaming format

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:[email protected]/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

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

Source: ONVIF

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:[email protected]
  reolink1: onvif://admin:[email protected]:8000
  tapo1: onvif://admin:[email protected]: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.

  • FFmpeg preistalled for Docker and Hass Add-on users
  • Hass Add-on users can target files from /media folder

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:[email protected]/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"
  • You can use go2rtc stream name as ffmpeg input (ex. ffmpeg:camera1#video=h264)
  • You can use video and audio params multiple times (ex. #video=copy#audio=copy#audio=pcmu)
  • You can use rotate param with 90, 180, 270 or -90 values, important with transcoding (ex. #video=h264#rotate=90)
  • You can use width and/or height params, important with transcoding (ex. #video=h264#width=1280)
  • You can use drawtext to add a timestamp (ex. drawtext=x=2:y=2:fontsize=12:fontcolor=white:box=1:boxcolor=black)
    • This will greatly increase the CPU of the server, even with hardware acceleration
  • You can use raw param for any additional FFmpeg arguments (ex. #raw=-vf transpose=1)
  • You can use input param to override default input template (ex. #input=rtsp/udp will change RTSP transport from TCP to UDP+TCP)
    • You can use raw input value (ex. #input=-timeout 5000000 -i {input})
    • You can add your own input templates

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.

  • check available devices in Web interface
  • video_size and framerate must be supported by your camera!
  • for Linux supported only video for now
  • for macOS you can stream Facetime camera or whole Desktop!
  • for macOS important to set right framerate

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 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.

The source can be used with:

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 -

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: HomeKit

Important:

  • You can use HomeKit Cameras without Apple devices (iPhone, iPad, etc.), it's just a yet another protocol
  • HomeKit device can be paired with only one ecosystem. So, if you have paired it to an iPhone (Apple Home) - you can't pair it with Home Assistant or go2rtc. Or if you have paired it to go2rtc - you can't pair it with iPhone
  • HomeKit device should be in same network with working mDNS between device and go2rtc

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:

  • HomeKit audio uses very non-standard AAC-ELD codec with very non-standard params and specification violation
  • Audio can be transcoded by ffmpeg source with #async option
  • Audio can be played by ffplay with -use_wallclock_as_timestamps 1 -async 1 options
  • Audio can't be played in VLC and probably any other player

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#async

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

Other names: ESeeCloud, dvr163.

  • you can skip username, password, port, ch and stream if they are default
  • setup separate streams for different channels and streams
streams:
  camera1: bubble://username:[email protected]:34567/bubble/live?ch=0&stream=0

Source: DVRIP

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

  • you can skip username, password, port, channel and subtype if they are default
  • setup separate streams for different channels
  • use subtype=0 for Main stream, and subtype=1 for Extra1 stream
  • only the TCP protocol is supported
streams:
  camera1: dvrip://username:[email protected]:34567?channel=0&subtype=0

Source: Tapo

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

  • stream quality is the same as RTSP protocol
  • use the cloud password, this is not the RTSP password! you do not need to add a login!
  • you can also use UPPERCASE MD5 hash from your cloud password with admin username
streams:
  # cloud password without username
  camera1: tapo://[email protected]
  # admin username and UPPERCASE MD5 cloud-password hash
  camera2: tapo://admin:[email protected]

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

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

The Nest API only allows you to get a link to a stream for 5 minutes. So every 5 minutes the stream will be reconnected.

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

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:[email protected]:554/Streaming/Channels/101
    - isapi://admin:[email protected]:80/

Source: Nest

Currently only WebRTC cameras are supported. Stream reconnects every 5 minutes.

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

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

  • Roborock S6 MaxV - only video (the vacuum has no microphone)
  • Roborock S7 MaxV - video and two way audio

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

This source type support four connection formats.

whep

WebRTC/WHEP - is an unapproved 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.

wyze

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

kinesis

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-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

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.

  • Go2rtc also can accepts incoming sources in RTSP, HTTP and WebRTC/WHIP formats
  • Go2rtc won't stop such a source if it has no clients
  • You can push data only to existing stream (create stream with empty source in config)
  • You can push multiple incoming sources to same stream
  • You can push data to non empty stream, so it will have additional codecs inside

Examples

  • RTSP with any codec 
    ffmpeg -re -i BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp rtsp://localhost:8554/camera1
  • HTTP-MJPEG with MJPEG codec 
    ffmpeg -re -i BigBuckBunny.mp4 -c mjpeg -f mpjpeg http://localhost:1984/api/stream.mjpeg?dst=camera1
  • HTTP-FLV with H264, AAC codecs 
    ffmpeg -re -i BigBuckBunny.mp4 -c copy -f flv http://localhost:1984/api/stream.flv?dst=camera1
  • MPEG-TS with H264 codec 
    ffmpeg -re -i BigBuckBunny.mp4 -c copy -f mpegts http://localhost:1984/api/stream.ts?dst=camera1

Incoming: Browser

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

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

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
  • you can stream: local files, web files, live streams or any format, supported by FFmpeg
  • you should use ffmpeg source for transcoding audio to codec, that your camera supports
  • you can check camera codecs on the go2rtc WebUI info page when the stream is active
  • some cameras support only low quality PCMA/8000 codec (ex. Tapo)
  • it is recommended to choose higher quality formats if your camera supports them (ex. PCMA/48000 for some Dahua cameras)
  • if you play files over http-link, you need to add #input=file params for transcoding, so file will be transcoded and played in real time
  • if you play live streams, you should skip #input param, because it is already in real time
  • you can stop active playback by calling the API with the empty src parameter
  • you will see one active producer and one active consumer in go2rtc WebUI info page during streaming

Module: API

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

API description.

Module config

  • 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: "127.0.0.1:1984" 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: ":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-----

PS:

  • MJPEG over WebSocket plays better than native MJPEG because Chrome bug
  • MP4 over WebSocket was created only for Apple iOS because it doesn't support MSE and native MP4

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:

  • default_query: "mp4" - MP4 compatible codecs (H264, H265, AAC)
  • default_query: "video=all&audio=all" - all tracks from all source (not all players can handle this)
  • default_query: "video=h264,h265" - only one video track (H264 or H265)
  • default_query: "video&audio=all" - only one first any video and all audio as separate tracks

Read more about codecs filters.

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.

  • by default, WebRTC uses both TCP and UDP on port 8555 for connections
  • you can use this port for external access
  • you can change the port in YAML config:
webrtc:
  listen: ":8555"  # address of your local server and port (TCP/UDP)

Static public IP

  • forward the port 8555 on your router (you can use same 8555 port or any other as external port)
  • add your external IP-address and external port to YAML config
webrtc:
  candidates:
    - 216.58.210.174:8555  # if you have static public IP-address

Dynamic public IP

  • forward the port 8555 on your router (you can use same 8555 port or any other as the external port)
  • add stun word and external port to YAML config
    • go2rtc automatically detects your external address with STUN-server
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: WebTorrent

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 situation when you have Internet with private IP-address.

  • Ngrok preistalled for Docker and Hass Add-on users
  • 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. 1984)
  • 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: 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

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 > rtsp://127.0.0.1:8554/camera1 (change to your stream name)

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:

  • MP4 snapshot: http://192.168.1.123:1984/api/frame.mp4?src=camera1 (H264, H265)
  • MP4 stream: http://192.168.1.123:1984/api/stream.mp4?src=camera1 (H264, H265, AAC)
  • MP4 file: http://192.168.1.123:1984/api/stream.mp4?src=camera1 (H264, H265*, AAC, OPUS, MP3, PCMA, PCMU, PCM)
    • You can use mp4, mp4=flac and mp4=all param for codec filters
    • You can use duration param in seconds (ex. duration=15)
    • You can use filename param (ex. filename=record.mp4)
    • You can use rotate param with 90, 180 or 270 values
    • You can use scale param with positive integer values (ex. scale=4:3)

Read more about codecs filters.

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

Module: HLS

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:

  • HLS/TS stream: http://192.168.1.123:1984/api/stream.m3u8?src=camera1 (H264)
  • HLS/fMP4 stream: http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4 (H264, H265, AAC)

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:

  • some cameras support MJPEG codec inside RTSP stream (ex. second stream for Dahua cameras)
  • some cameras has HTTP link with MJPEG stream
  • some cameras has HTTP link with snapshots - go2rtc can convert them to MJPEG stream
  • you can convert H264/H265 stream from your camera via FFmpeg integraion

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

streams:
  camera1:
    - rtsp://rtsp:[email protected]/av_stream/ch0
    - ffmpeg:camera1#video=mjpeg

API examples:

  • MJPEG stream: http://192.168.1.123:1984/api/stream.mjpeg?src=camera1
  • JPEG snapshots: http://192.168.1.123:1984/api/frame.jpeg?src=camera1
    • You can use width/w and/or height/h params
    • You can use rotate param with 90, 180, 270 or -90 values
    • You can use hardware/hw param 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
  • local access to RTSP is not a problem for FFmpeg integration, because it runs locally on your server
  • local access to API is not a problem for Home Assistant Add-on, because Hass runs locally on same server and Add-on Web UI protected with Hass authorization (Ingress feature)
  • external access to WebRTC TCP port is not a problem, because it used only for transmit encrypted media data
    • anyway you need to open this port to your local network and to the Internet in order for WebRTC to work

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:

  • RTSP will provide only the first video and only the first audio (any codec)
  • MP4 will include only compatible codecs (H264, H265, AAC)
  • HLS will output in the legacy TS format (H264 without audio)

Some examples:

  • rtsp://192.168.1.123:8554/camera1?mp4 - useful for recording as MP4 files (e.g. Hass or Frigate)
  • rtsp://192.168.1.123:8554/camera1?video=h264,h265&audio=aac - full version of the filter above
  • rtsp://192.168.1.123:8554/camera1?video=h264&audio=aac&audio=opus - H264 video codec and two separate audio tracks
  • rtsp://192.168.1.123:8554/camera1?video&audio=all - any video codec and all audio codecs as separate tracks
  • http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4 - HLS stream with MP4 compatible codecs (HLS/fMP4)
  • http://192.168.1.123:1984/api/stream.m3u8?src=camera1&mp4=flac - HLS stream with PCMA/PCMU/PCM audio support (HLS/fMP4), won't work on old devices
  • http://192.168.1.123:1984/api/stream.mp4?src=camera1&mp4=flac - MP4 file with PCMA/PCMU/PCM audio support, won't work on old devices (ex. iOS 12)
  • http://192.168.1.123:1984/api/stream.mp4?src=camera1&mp4=all - MP4 file with non standard audio codecs, won't work on some players

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+ H264, OPUS, PCMU, PCMA H264, H265*, AAC, FLAC*, OPUS H264, H265*, AAC, FLAC*, OPUS, MP3 no
Desktop Edge H264, OPUS, PCMU, PCMA H264, H265*, AAC, FLAC*, OPUS H264, H265*, AAC, FLAC*, OPUS, MP3 no
Android Chrome 107+ H264, OPUS, PCMU, PCMA H264, H265*, AAC, FLAC*, OPUS H264, H265*, AAC, FLAC*, OPUS, MP3 no
Desktop Firefox H264, OPUS, PCMU, PCMA H264, AAC, FLAC*, OPUS H264, AAC, FLAC*, OPUS no
Desktop Safari 14+ H264, H265*, OPUS, PCMU, PCMA H264, H265, AAC, FLAC* no! H264, H265, AAC, FLAC*
iPad Safari 14+ H264, H265*, OPUS, PCMU, PCMA H264, H265, AAC, FLAC* no! H264, H265, AAC, FLAC*
iPhone Safari 14+ H264, H265*, OPUS, PCMU, PCMA 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

  • Chrome H265: read this and read this
  • Edge H265: read this
  • Desktop Safari H265: Menu > Develop > Experimental > WebRTC H265
  • iOS Safari H265: Settings > Safari > Advanced > Experimental > WebRTC H265

Audio

  • Go2rtc support automatic repack PCMA/PCMU/PCM codecs to FLAC for MSE/MP4/HLS so they will work almost anywhere
  • WebRTC audio codecs: PCMU/8000, PCMA/8000, OPUS/48000/2
  • OPUS and MP3 inside MP4 is part of the standard, but some players do not support them anyway (especially Apple)

Apple devices

  • all Apple devices don't support HTTP progressive streaming
  • iPhones don't support MSE technology because it competes with the HTTP Live Streaming (HLS) technology, invented by Apple
  • HLS is the worst technology for live streaming, it still exists only because of iPhones

Codec names

  • H264 = H.264 = AVC (Advanced Video Coding)
  • H265 = H.265 = HEVC (High Efficiency Video Coding)
  • PCMU = G.711 PCM (A-law) = PCM A-law (alaw)
  • PCMA = G.711 PCM (µ-law) = PCM mu-law (mulaw)
  • PCM = L16 = PCM signed 16-bit big-endian (s16be)
  • AAC = MPEG4-GENERIC
  • MP3 = MPEG-1 Audio Layer III or MPEG-2 Audio Layer III

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

  • FLAC codec not supported in a RTSP stream. If you using Frigate or Hass for recording MP4 files with PCMA/PCMU/PCM audio - you should setup transcoding to AAC codec.
  • PCMA and PCMU are VERY low quality codecs. Them support only 256! different sounds. Use them only when you have no other options.

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:

streams:
  dahua:
    - rtsp://admin:[email protected]/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
    - ffmpeg:rtsp://admin:[email protected]/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

Cameras experience

  • Dahua - reference implementation streaming protocols, a lot of settings, high stream quality, multiple streaming clients
  • EZVIZ - awful RTSP protocol realisation, many bugs in SDP
  • Hikvision - a lot of proprietary streaming technologies
  • Reolink - some models has awful unusable RTSP realisation and not best RTMP alternative (I recommend that you contact Reolink support for new firmware), few settings
  • Sonoff - very low stream quality, no settings, not best protocol implementation
  • TP-Link - few streaming clients, packet loss?
  • Chinese cheap noname cameras, Wyze Cams, Xiaomi cameras with hacks (usual has /live/ch00_1 in RTSP URL) - awful but usable RTSP protocol realisation, low stream quality, few settings, packet loss?

TIPS

Using apps for low RTSP delay

  • ffplay -fflags nobuffer -flags low_delay "rtsp://192.168.1.123:8554/camera1"
  • VLC > Preferences > Input / Codecs > Default Caching Level: Lowest Latency

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.

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