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, HTTP (FLV/MJPEG/JPEG), FFmpeg, USB Cameras and other sources
- streaming to RTSP, WebRTC, MSE/MP4 or MJPEG
- first project in the World with support streaming from HomeKit Cameras
- first project in the World with support H265 for WebRTC in browser (Safari only, read more)
- on the fly transcoding for unsupported codecs via FFmpeg
- multi-source 2-way codecs negotiation
- mixing tracks from different sources to single stream
- auto match client supported codecs
- 2-way audio for
ONVIF Profile T
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
- Fast start
- Configuration
- Security
- Codecs madness
- Codecs negotiation
- TIPS
- FAQ
- Download binary or use Docker or Home Assistant Add-on
- 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_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_mipsel
- Linux MIPS (ex. Xiaomi Gateway 3)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.
- 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
Container alexxit/go2rtc with support amd64
, 386
, arm64
, arm
. This container is the same as Home Assistant Add-on, but can be used separately from Home Assistant. Container has preinstalled FFmpeg, Ngrok and Python.
- by default go2rtc will search
go2rtc.yaml
in the current work dirrectory api
server will start on default 1984 port (TCP)rtsp
server will start on default 8554 port (TCP)webrtc
will use port 8555 (TCP/UDP) for 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
- 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 - rtmp -
RTMP
streams - http -
HTTP-FLV
,JPEG
(snapshots),MJPEG
streams - ffmpeg - FFmpeg integration (
HLS
,files
and many others) - ffmpeg:device - local USB Camera or Webcam
- exec - advanced FFmpeg and GStreamer integration
- echo - get stream link from bash or python
- homekit - streaming from HomeKit Camera
- ivideon - public cameras from Ivideon service
- hass - Home Assistant integration
- Support RTSP and RTSPS links with multiple video and audio tracks
- Support 2-way audio ONLY for ONVIF Profile T cameras (back channel connection)
Attention: other 2-way audio standards are not supported! ONVIF without Profile T is not supported!
streams:
sonoff_camera: rtsp://rtsp:[email protected]/av_stream/ch0
If your camera has two RTSP links - you can add both of them as sources. This is useful when streams has different codecs, as example AAC audio with main stream and PCMU/PCMA audio with second stream.
Attention: Dahua cameras has different capabilities for different RTSP links. For example, it has support multiple codecs for 2-way audio with &proto=Onvif
in link and only one codec without it.
streams:
dahua_camera:
- rtsp://admin:[email protected]/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
- rtsp://admin:[email protected]/cam/realmonitor?channel=1&subtype=1
PS. For disable bachannel just add #backchannel=0
to end of RTSP link.
You can get stream from RTMP server, for example Frigate.
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
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
PS. Dahua camera has bug: if you select MJPEG codec for RTSP second stream - snapshot won't work.
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
, h264/ultra
, h264/high
, 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 support ffmpeg..."
- You can use
video
andaudio
params multiple times (ex.#video=copy#audio=copy#audio=pcmu
) - You can use go2rtc stream name as ffmpeg input (ex.
ffmpeg:camera1#video=h264
) - You can use
rotate
params 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
raw
param for any additional FFmpeg arguments (ex.#raw=-vf transpose=1
).
Read more about encoding hardware acceleration.
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
resolution
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
streams:
linux_usbcam: ffmpeg:device?video=0&resolution=1280x720#video=h264
windows_webcam: ffmpeg:device?video=0#video=h264
macos_facetime: ffmpeg:device?video=0&audio=1&resolution=1280x720&framerate=30#video=h264#audio=pcma
FFmpeg source just a shortcut to exec source. You can get any stream or file or device via FFmpeg or GStreamer and push it to go2rtc via RTSP protocol:
streams:
stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
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
Important:
- You can use HomeKit Cameras without Apple devices (iPhone, iPad, etc.), it's just a yet another protocol
- HomeKit device can be paired with only one ecosystem. So, if you have paired it to an iPhone (Apple Home) - you can't pair it with Home Assistant or go2rtc. Or if you have paired it to go2rtc - you can't pair it with iPhone
- HomeKit device should be in same network with working mDNS between device and go2rtc
go2rtc support import paired HomeKit devices from Home Assistant. So you can use HomeKit camera with Hass and go2rtc simultaneously. If you using Hass, I recommend pairing devices with it, it will give you more options.
You can pair device with go2rtc on the HomeKit page. If you can't see your devices - reload the page. Also try reboot your HomeKit device (power off). If you still can't see it - you have a problems with mDNS.
If you see a device but it does not have a pair button - it is paired to some ecosystem (Apple Home, Home Assistant, HomeBridge etc). You need to delete device from that ecosystem, and it will be available for pairing. If you cannot unpair device, you will have to reset it.
Important:
- HomeKit audio uses very non-standard AAC-ELD codec with very non-standard params and specification violation
- Audio can be transcoded by ffmpeg source with
#async
option - Audio can be played by
ffplay
with-use_wallclock_as_timestamps 1 -async 1
options - Audio can't be played in
VLC
and probably any other player
Recommended settings for using HomeKit Camera with WebRTC, MSE, MP4, RTSP:
streams:
aqara_g3:
- hass:Camera-Hub-G3-AB12
- ffmpeg:aqara_g3#audio=aac#audio=opus#async
RTSP link with "normal" audio for any player: rtsp://192.168.1.123:8554/aqara_g3?video&audio=aac
This source is in active development! Tested only with Aqara Camera Hub G3 (both EU and CN versions).
Support public cameras from service Ivideon.
streams:
quailcam: ivideon:100-tu5dkUPct39cTp9oNEN2B6/0
Support import camera links from Home Assistant config files:
- support Generic Camera, setup via GUI
- support HomeKit 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
More cameras, like Tuya, ONVIF, and possibly others can also be imported by using this method.
The HTTP API is the main part for interacting with the application. Default address: http://127.0.0.1:1984/
.
go2rtc has its own JS video player (video-rtc.js
) with:
- support technologies:
- WebRTC over UDP or TCP
- MSE or MP4 or MJPEG over WebSocket
- automatic selection best technology according on:
- codecs inside your stream
- current browser capabilities
- current network configuration
- automatic stop stream while browser or page not active
- automatic stop stream while player not inside page viewport
- automatic reconnection
Technology selection based on priorities:
- Video and Audio better than just Video
- H265 better than H264
- WebRTC better than MSE, than MP4, than MJPEG
go2rtc has simple HTML page (stream.html
) with support params in URL:
- multiple streams on page
src=camera1&src=camera2...
- stream technology autoselection
mode=webrtc,mse,mp4,mjpeg
- stream technology comparison
src=camera1&mode=webrtc&mode=mse&mode=mp4
- player width setting in pixels
width=320px
or percentswidth=50%
Module config
- you can disable HTTP API with
listen: ""
and use, for example, only RTSP client/server protocol - you can enable HTTP API only on localhost with
listen: "127.0.0.1:1984"
setting - you can change API
base_path
and host go2rtc on your main app webserver suburl - all files from
static_dir
hosted on root path:/
api:
listen: ":1984" # default ":1984", HTTP API port ("" - disabled)
username: "admin" # default "", Basic auth for WebUI
password: "pass" # default "", Basic auth for WebUI
base_path: "/rtc" # default "", API prefix for serve on suburl (/api => /rtc/api)
static_dir: "www" # default "", folder for static files (custom web interface)
origin: "*" # default "", allow CORS requests (only * supported)
PS:
- go2rtc doesn't provide HTTPS. Use Nginx or Ngrok or Home Assistant Add-on for this tasks
- you can access microphone (for 2-way audio) only with HTTPS (read more)
- 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 omit the codec filters, so one first video and one first audio will be selected
- you can set
?video=copy
or just?video
, so only one first video without audio will be selected - you can set multiple video or audio, so all of them will be selected
- 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
WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of Internet do you have.
- by default, WebRTC 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
With Ngrok integration you can get external access to your streams in situation when you have Internet with private IP-address.
- Ngrok preistalled for Docker and Hass Add-on users
- you may need external access for two different things:
- WebRTC stream, so you need tunnel WebRTC TCP port (ex. 8555)
- go2rtc web interface, so you need tunnel API HTTP port (ex. 1984)
- Ngrok support authorization for your web interface
- Ngrok automatically adds HTTPS to your web interface
Ngrok free subscription limitations:
- you will always get random external address (not a problem for webrtc stream)
- you can forward multiple ports but use only one Ngrok app
go2rtc will automatically get your external TCP address (if you enable it in ngrok config) and use it with WebRTC connection (if you enable it in webrtc config).
You need manually download Ngrok agent app for your OS and register in Ngrok service.
Tunnel for only WebRTC Stream
You need to add your Ngrok token and WebRTC TCP port to YAML:
ngrok:
command: ngrok tcp 8555 --authtoken eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
Tunnel for WebRTC and Web interface
You need to create ngrok.yaml
config file and add it to go2rtc config:
ngrok:
command: ngrok start --all --config ngrok.yaml
Ngrok config example:
version: "2"
authtoken: eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
tunnels:
api:
addr: 1984 # use the same port as in go2rtc config
proto: http
basic_auth:
- admin:password # you can set login/pass for your web interface
webrtc:
addr: 8555 # use the same port as in go2rtc config
proto: tcp
If you install go2rtc as Hass Add-on - you need to use localhost IP-address. In other cases you need to use IP-address of server with go2rtc application.
Add any supported stream source as Generic Camera and view stream with built-in Stream integration. Technology HLS
, supported codecs: H264
, poor latency.
- Add your stream to go2rtc config
- Hass > Settings > Integrations > Add Integration > Generic Camera >
rtsp://127.0.0.1:8554/camera1
View almost any Hass camera using WebRTC
technology, supported codecs H264
/PCMU
/PCMA
/OPUS
, best latency.
When the stream starts - the camera entity_id
will be added to go2rtc "on the fly". You don't need to add cameras manually to go2rtc config. Some cameras (like Nest) have a dynamic link to the stream, it will be updated each time a stream is started from the Hass interface.
- 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
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.
Provides several features:
- MSE stream (fMP4 over WebSocket)
- Camera snapshots in MP4 format (single frame), can be sent to Telegram
- MP4 "file stream" - bad format for streaming because of high start delay, doesn't work in Safari
API examples:
- MP4 stream:
http://192.168.1.123:1984/api/stream.mp4?src=camera1
- MP4 snapshot:
http://192.168.1.123:1984/api/frame.mp4?src=camera1
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 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.
AVC/H.264
codec 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 | MP4 |
---|---|---|---|
latency | best | medium | bad |
Desktop Chrome 107 | H264 | H264, H265* | H264, H265* |
Desktop Edge | H264 | H264, H265* | H264, H265* |
Desktop Safari | H264, H265* | H264, H265 | no! |
Desktop Firefox | H264 | H264 | H264 |
Android Chrome 107 | H264 | H264, H265* | H264 |
iPad Safari 13 | H264, H265* | H264, H265 | no! |
iPhone Safari 13 | H264, H265* | no! | no! |
masOS Hass App | no | no | no |
- 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
- WebRTC audio codecs:
PCMU/8000
,PCMA/8000
,OPUS/48000/2
- MSE/MP4 audio codecs:
AAC
Apple devices
- all Apple devices don't support MP4 stream (they only support progressive loading of static files)
- iPhones don't support MSE technology because it competes with the HLS technology, invented by Apple
- HLS is the worst technology for live streaming, it still exists only because of iPhones
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.
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. Why go2rtc is an addon and not an integration?
Because go2rtc is more than just viewing your stream online with WebRTC. 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.
When go2rtc is released, the WebRTC Camera integration will be updated. And you can decide whether to use the integration or the 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 protocol.
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.