GitHub - AlexxIT/go2rtc: Ultimate camera streaming application with support RTSP, RTMP, HTTP-FLV, WebRTC, MSE, HLS, MP4, MJPEG, HomeKit, FFmpeg, etc.

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

zero-dependency and zero-config small app for all OS (Windows, macOS, Linux, ARM)
zero-delay for many supported protocols (lowest possible streaming latency)
streaming from RTSP, RTMP, DVRIP, HTTP (FLV/MJPEG/JPEG/TS), USB Cameras and other sources
streaming from any sources, supported by FFmpeg
streaming to RTSP, WebRTC, MSE/MP4, HomeKit HLS or MJPEG
publish any source to popular streaming services (YouTube, Telegram, etc.)
first project in the World with support streaming from HomeKit Cameras
support H265 for WebRTC in browser (Safari only, read more)
on the fly transcoding for unsupported codecs via FFmpeg
play audio files and live streams on some cameras with speaker
multi-source 2-way codecs negotiation
- mixing tracks from different sources to single stream
- auto match client supported codecs
- 2-way audio for some cameras
streaming from private networks via ngrok
can be integrated to any smart home platform or be used as standalone app

Inspired by:

series of streaming projects from @deepch
webrtc go library and whole @pion team
rtsp-simple-server idea from @aler9
GStreamer framework pipeline idea
MediaSoup framework routing idea
HomeKit Accessory Protocol from @brutella
creator of the project's logo @v_novoseltsev

Fast start
Configuration
Security
Codecs filters
Codecs madness
Codecs negotiation
Projects using go2rtc
Camera experience
TIPS
FAQ

Fast start

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

Optionally:

add your streams to config file
setup external access to webrtc

Developers:

write your own web interface
integrate web api into your smart home platform

go2rtc: Binary

Download binary for your OS from latest release:

go2rtc_win64.zip - Windows 10+ 64-bit
go2rtc_win32.zip - Windows 7+ 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 - macOS 10.13+ Intel 64-bit
go2rtc_mac_arm64.zip - macOS ARM 64-bit

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

Install Add-On:
- Settings > Add-ons > Plus > Repositories > Add https://github.com/AlexxIT/hassio-addons
- go2rtc > Install > Start
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: latest nightly release
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 directory
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
expr - get stream link via built-in expression language
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
kasa - TP-Link Kasa cameras
gopro - GoPro cameras
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

Two way audio

Supported for sources:

RTSP cameras with ONVIF Profile T (back channel connection)
DVRIP cameras
TP-Link Tapo cameras
Hikvision ISAPI cameras
Roborock vacuums models with cameras
Exec audio on server
Any Browser as IP-camera

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:rtsp://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

Other options

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

Add custom timeout #timeout=30 (in seconds)
Ignore audio - #media=video or ignore video - #media=audio
Ignore two way audio API #backchannel=0 - important for some glitchy cameras
Use WebSocket transport #transport=ws...

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 Nginx with nginx-rtmp-module.

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

  # 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:[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

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 (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:

FFmpeg - go2rtc ffmpeg source just a shortcut to exec source
FFplay - play audio on your server
GStreamer
Raspberry Pi Cameras
any your own software

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

killsignal - signal which will be send to stop the process (numeric form)
killtimeout - time in seconds for forced termination with sigkill
backchannel - enable backchannel for two-way audio

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:

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't be played in VLC and probably any other player
Audio should be transcoded for using with MSE, WebRTC, etc.

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.

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

New in v1.2.0

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:
  only_stream: dvrip://username:[email protected]:34567?channel=0&subtype=0
  only_tts: dvrip://username:[email protected]:34567?backchannel=1
  two_way_audio:
    - dvrip://username:[email protected]:34567?channel=0&subtype=0
    - dvrip://username:[email protected]:34567?backchannel=1

Source: Tapo

New in v1.2.0

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
some new camera firmwares requires SHA256 instead of MD5

streams:
  # cloud password without username
  camera1: tapo://[email protected]
  # admin username and UPPERCASE MD5 cloud-password hash
  camera2: tapo://admin:[email protected]
  # admin username and UPPERCASE SHA256 cloud-password hash
  camera3: tapo://admin:[email protected]

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.

username - urlsafe email, [email protected] -> alex%40gmail.com
password - base64password, secret1 -> c2VjcmV0MQ==

streams:
  kc401: kasa://username:[email protected]: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:

Generic Camera, setup via GUI
HomeKit Camera
ONVIF
Roborock vacuums with camera

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

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

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.

Go2rtc also can accepts incoming sources in RTSP, RTMP, 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

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:

Create empty stream in the go2rtc.yaml
Go to go2rtc WebUI
Open links page for you stream
Select camera+microphone or display+speaker option
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:

Settings > Stream > Service: WHIP > http://192.168.1.123:1984/api/webrtc?dst=camera1

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

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

Publish stream

New in v1.8.0

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

Supported codecs: H264 for video and AAC for audio
AAC audio is required for YouTube, videos without audio will not work
You don't need to enable RTMP module listening for this task

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:[email protected]/stream1#video=h264#hardware#audio=aac
  audio_transcode:
    - ffmpeg:rtsp://user:[email protected]/stream1#video=copy#audio=aac

Telegram Desktop App > Any public or private channel or group (where you admin) > Live stream > Start with... > Start streaming.
YouTube > Create > Go live > Stream latency: Ultra low-latency > Copy: Stream URL + Stream key.

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

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: /
you can use raw TLS cert/key content or path to files

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:

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

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.

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

setup integration with ngrok service

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:

export any H264 camera to Apple HomeKit
transparent proxy any Apple HomeKit camera (Aqara, Eve, Eufy, etc.) back to Apple HomeKit, so you will have all camera features in Apple Home and also will have RTSP/WebRTC/MP4/etc. from your HomeKit camera

Important

HomeKit cameras supports only H264 video and OPUS audio

Minimal config

streams:
  dahua1: rtsp://admin:[email protected]/cam/realmonitor?channel=1&subtype=0
homekit:
  dahua1:  # same stream ID from streams list, default PIN - 19550224

Full config

streams:
  dahua1:
    - rtsp://admin:[email protected]/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

Video stream from HomeKit camera to Apple device (iPhone, AppleTV) will be transmitted directly
Video stream from HomeKit camera to RTSP/WebRTC/MP4/etc. will be transmitted via go2rtc

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:

Share any local stream via WebTorrent technology
Get any incoming stream from PC or mobile via WebTorrent technology
Get any remote go2rtc source via WebTorrent technology

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.

ngrok is pre-installed 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

The ngrok free subscription has the following limitations:

You can reserve a free domain for serving the web interface, but the TCP address you get will always be random and change with each restart of the ngrok agent (not a problem for webrtc stream)
You can forward multiple ports from a single agent, but you can only run one ngrok agent on the free plan

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:

Camera RTSP source => Generic Camera
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:

Camera Entity => Picture Entity Card => Technology HLS, codecs: H264/H265/AAC, poor latency.
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
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:

MSE stream (fMP4 over WebSocket)
Camera snapshots in MP4 format (single frame), can be sent to Telegram
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)

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:

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)

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

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

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+ - 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

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)
PCMA = G.711 PCM (A-law) = PCM A-law (alaw)
PCMU = 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

Frigate 12+ - open source NVR built around real-time AI object detection
Frigate Lovelace Card - custom card for Home Assistant
OpenIPC - Alternative IP Camera firmware from an open community
wz_mini_hacks - Custom firmware for Wyze cameras
EufyP2PStream - A small project that provides a Video/Audio Stream from Eufy cameras that don't directly support RTSP
ioBroker.euSec - ioBroker adapter for control Eufy security devices
MMM-go2rtc - MagicMirror² Module
ring-mqtt - Ring devices to MQTT Bridge

Distributions

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

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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assets		assets
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internal		internal
pkg		pkg
scripts		scripts
website		website
www		www
.gitignore		.gitignore
Dockerfile		Dockerfile
LICENSE		LICENSE
README.md		README.md
go.mod		go.mod
go.sum		go.sum
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main.go		main.go
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AlexxIT/go2rtc

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Fast start

go2rtc: Binary

go2rtc: Docker

go2rtc: Home Assistant Add-on

go2rtc: Home Assistant Integration

go2rtc: Dev version

Configuration

Module: Streams

Two way audio

Source: RTSP

Source: RTMP

Source: HTTP

Source: ONVIF

Source: FFmpeg

Source: FFmpeg Device

Source: Exec

Source: Echo

Source: Expr

Source: HomeKit

Source: Bubble

Source: DVRIP

Source: Tapo

Source: Kasa

Source: GoPro

Source: Ivideon

Source: Hass

Source: ISAPI

Source: Nest

Source: Roborock

Source: WebRTC

Source: WebTorrent

Incoming sources

Incoming: Browser

Incoming: WebRTC/WHIP

Stream to camera

Publish stream

Module: API

Module: RTSP

Module: RTMP

Module: WebRTC

Module: HomeKit

Module: WebTorrent

Module: ngrok

Module: Hass

Module: MP4

Module: HLS

Module: MJPEG

Module: Log

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