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
- 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
- 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
- Module: Streams
- Security
- Codecs filters
- Codecs madness
- Codecs negotiation
- Projects using go2rtc
- Camera experience
- TIPS
- FAQ
- 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
Download binary for your OS from latest release:
go2rtc_win64.zip
- Windows 64-bitgo2rtc_win32.zip
- Windows 32-bitgo2rtc_win_arm64.zip
- Windows ARM 64-bitgo2rtc_linux_amd64
- Linux 64-bitgo2rtc_linux_i386
- Linux 32-bitgo2rtc_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-bitgo2rtc_mac_arm64.zip
- Mac ARM 64-bit
Don't forget to fix the rights chmod x go2rtc_xxx_xxx
on Linux and Mac.
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.
- Install Add-On:
- Settings > Add-ons > Plus > Repositories > Add
https://github.com/AlexxIT/hassio-addons
- go2rtc > Install > Start
- Settings > Add-ons > Plus > Repositories > Add
- Setup 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.
Latest, but maybe unstable version:
- Binary: latest nightly release
- Docker:
alexxit/go2rtc:master
oralexxit/go2rtc:master-hardware
versions - Hass Add-on:
go2rtc master
orgo2rtc master hardware
versions
- 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 connectionsffmpeg
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
go2rtc support different stream source types. You can config one or multiple links of any type as stream source.
Available source types:
- rtsp -
RTSP
andRTSPS
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 usingONVIF
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
Read more about incoming sources
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.
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 ofrtsps://
. 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
You can get stream from RTMP server, for example Nginx with nginx-rtmp-module.
streams:
rtmp_stream: rtmp://192.168.1.123/live/camera1
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.
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
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
andaudio
params multiple times (ex.#video=copy#audio=copy#audio=pcmu
) - You can use
rotate
param with90
,180
,270
or-90
values, important with transcoding (ex.#video=h264#rotate=90
) - You can use
width
and/orheight
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
- You can use raw input value (ex.
Read more about hardware acceleration.
PS. It is recommended to check the available hardware in the WebUI add page.
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
andframerate
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.
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 sigkillbackchannel
- 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
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
Like echo
source, but uses the built-in expr expression language (read more).
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).
Other names: ESeeCloud, dvr163.
- you can skip
username
,password
,port
,ch
andstream
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
Other names: DVR-IP, NetSurveillance, Sofia protocol (NETsurveillance ActiveX plugin XMeye SDK).
- you can skip
username
,password
,port
,channel
andsubtype
if they are default - setup separate streams for different channels
- use
subtype=0
for Main stream, andsubtype=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
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)}'
TP-Link Kasa non-standard protocol more info.
username
- urlsafe email,[email protected]
->[email protected]
password
- base64password,secret1
->c2VjcmV0MQ==
streams:
kc401: kasa://username:[email protected]:19443/https/stream/mixed
Tested: KD110, KC200, KC401, KC420WS, EC71.
Support streaming from GoPro cameras, connected via USB or Wi-Fi to Linux, Mac, Windows. Read more.
Support public cameras from service Ivideon.
streams:
quailcam: ivideon:100-tu5dkUPct39cTp9oNEN2B6/0
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.
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/
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=***
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.
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.
This source can get a stream from another go2rtc via WebTorrent protocol.
streams:
webtorrent1: webtorrent:?share=huofssuxaty00izc&pwd=k3l2j9djeg8v8r7e
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
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
ordisplay speaker
option - Open
webrtc
local page (your go2rtc should work over HTTPS!) orshare link
via WebTorrent technology (work over HTTPS by default)
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
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
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.
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.
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
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.
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!
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
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
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...
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.
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
=> TechnologyHLS
, codecs:H264/H265/AAC
, poor latency.Camera Entity
=> RTSPtoWebRTC =>Picture Entity Card
=> TechnologyWebRTC
, 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
orCamera 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.
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
andmp4=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 with90
,180
or270
values - You can use
scale
param with positive integer values (ex.scale=4:3
)
- You can use
Read more about codecs filters.
PS. Rotate and scale params don't use transcoding and change video using metadata.
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.
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/orheight
/h
params - You can use
rotate
param with90
,180
,270
or-90
values - You can use
hardware
/hw
param read more
- You can use
PS. This module also supports streaming to the server console (terminal) in the animated ASCII art format (read more):
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
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.
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 abovertsp://192.168.1.123:8554/camera1?video=h264&audio=aac&audio=opus
- H264 video codec and two separate audio tracksrtsp://192.168.1.123:8554/camera1?video&audio=all
- any video codec and all audio codecs as separate trackshttp://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 deviceshttp://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
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 toFLAC
for MSE/MP4/HLS so they will work almost anywhere - WebRTC audio codecs:
PCMU/8000
,PCMA/8000
,OPUS/48000/2
OPUS
andMP3
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
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.
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 thePCMU/8000
orPCMA/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.
- 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
- 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?
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
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.