This error not related to this component
Iām trying to sync the volume, so I can remotely adjust it when casting, but I have 2 challenges:
- I canāt seem to unmute the card, the default says muted all the time (after unmuting, setting the volume works fine)
- theme: Backend-selected
title: Camera HomeFlex
path: camera-homeflex
badges:
cards:
- type: custom:webrtc-camera
url: >-
ffmpeg:rtsp://XXXX:[email protected]/live0#video=copy#audio=opus
media: video,audio
mode: webrtc
muted: false
background: true- type: entities
entities:
- input_number.homeflex_volume
- type: custom:content-card-volume
entity: input_number.homeflex_volume
- my javascript code to find the volume element (custom:content-card-volume) is extremely dirty, is there a unique ID I could hook into? It works once unmuted ā¦ but this is not sustainable code
try { //WebRTC layout var b = document.querySelector("body > home-assistant") .shadowRoot.querySelector("home-assistant-main") .shadowRoot.querySelector("ha-drawer > partial-panel-resolver > ha-panel-lovelace") .shadowRoot.querySelector("hui-root").shadowRoot.querySelector("#view > hui-view > hui-masonry-view") .shadowRoot.querySelector("#columns > div:nth-child(1) > webrtc-camera") .shadowRoot.querySelector("ha-card > div.player > div > video") b.muted = false b.volume = newVol } catch (error) { }
For anyone else looking for this:
style: '.header {display: none} .pictureinpicture {display: none}'
@RT1080 Thanks for posting that. Iām trying to do the same with multiple style instructions, however the pictureinpicture style still isnāt applying - in fact, only one of my intended styles is applying, and itās the .mode one. The .mode style isnāt the first in my list either, but itās in the middle of the other styles that I am trying to set:
style: 'video {object-fit: fill} .pictureinpicture {display: none} .mode {display: none} .fullscreen {display: none}'
any ideas what might be preventing the others from applying?
Below works for meā¦
style: >-
video {object-fit: fill;} .header {display: none} .pictureinpicture {display: none} .mode {display: none} .fullscreen {display: none}'
Has your WebRTC actually been integrated into the latest version of Frigate?
Somehow it is not clear to me why the Go2rtc log from Frigate shows the following (as an example):
15:59:30.580 WRN github.com/AlexxIT/go2rtc/cmd/streams/producer.go:132 > error=āread tcp 10.1.1.41:59958->10.1.1.60:554: i/o timeoutā url=rtsp://admin:[email protected]:554/live/ch0
Although I have not installed WebRTC from you. Or I used to have it. But then I uninstalled it because I thought it was integrated into Frigate by default.
In older docs it says you can install it/ or need it for WebRTC, in other newer ones it is no longer necessaryā¦
Itās getting more and more confusing.
the same with this config:
webrtc:
candidates:
- INTERNAL-HA-IP:8555
- stun:8555
On the one hand it says in docs you should add this in frigate.yaml for smooth function. On the other hand, it says that this is no longer necessary with the latest Frigate version.
There is a frigate forum/git. Youāre better off discussing there but I donāt think they have the latest versions.
Trying to get audio working from a couple older cameras. Only codec options on the cameras are AAC, LPCM, G.711, and G.726. Am I correct that itās not possible to get audio working with WebRTC?
AAC audio works without using WebRTC but the stream delay is horribleā¦ maybe 10-15 seconds behind. Video only with WebRTC works fantastic with almost no delay. Really want that audio thoughā¦
Hi James - not sure if you are stil using these forums, but figured Iād give it a shotā¦ I also have a Nest Doorbell, and while I can use the below code to show the live stream, it appears to revert to a static image at some point, but I don;t know how much time passes before this occurs as I canāt tell its become static until I try and dance around in front of it when my wife can confirm if she can see me or not.
type: grid
square: false
columns: 1
cards:
- show_state: true
show_name: true
camera_view: live
type: picture-entity
entity: camera.basement_doorbell
camera_image: camera.basement_doorbell
I havenāt been able to program the camera to work with WebRTC as it asks me for a URL of the camera and Iām not sure where to find this information, can you give me a bit more info on how your parents dashboard is setup in this regard?
This was the only way I could get my Foscam interior cameras to work with Home Assistant. I did end up having to go into the device configs in /.storage and change the RTSP port to the same as the IP port (port 88).
That said - the feed works perfectly fine in realtime if Iām on wifi, but shows the play-symbol with a line through it when Iām on mobile data. It just never loads. Itās falling back to MSE in the top right corner. My speed should be fine and I donāt have any issues streaming things from mobile data. Itās also streaming the 720p sub stream.
This is one of my cards (sensitive info removed):
type: custom:webrtc-camera
url: rtsp://username:password@IP:port/videoSub
mode: webrtc,webrtc/tcp,mse,hls,mjpeg
media: video,audio
Good evening everyone, please tell me I have about 10 cameras, are they all connected to go2rtc via rtsp? How can they all be connected to a zero channel and get an rtsp link?
Dont know if this will help you. This is my setup in HA addon.
rtsp:
listen: ā:8554ā
default_query: mp4
streams:
CAM1: rtsp://USERNAME:PASSWORD@cameraIP:8554/live0.264
CAM2: rtsp://USERNAME:PASSWORD@cameraIP:8554/live1.264
CAM3: rtsp://USERNAME:PASSWORD@cameraIP:8554/live0.264
(These are for Holowits cameras)
rtsp://GO2RTCserverIP:8554/CAM1?mp4
or
rtsp://GO2RTCserverIP:8554/CAM1?video=all&audio=all
or
rtsp://GO2RTCserverIP:8554/CAM1
The rtsp link becomes the name that you specify
Hello everyone,
I canāt make 2-ways audio work.
All I can do is Hearing the audio from my browser. But I canāt send audio from microphone to camera. (my camera brand is EZVIZ C6N which is support 2-ways talk)
Could anyone helps?
Here is my go2rtc config section (inside frigate):
ffmpeg:
hwaccel_args: preset-rpi-64-h264
go2rtc:
streams:
CameraEZVIZC6N:
- rtsp://USER:[email protected]:554/live0
- "ffmpeg:CameraEZVIZC6N#audio=opus"
- "ffmpeg:CameraEZVIZC6N#audio=pcmu"
- "ffmpeg:CameraEZVIZC6N#audio=pcma"
- "ffmpeg:CameraEZVIZC6N#audio=aac"
rtsp:
listen: ":8554"
webrtc:
candidates:
- 192.168.1.128:8555
- stun:8555
log:
level: debug
api: debug
rtsp: debug
streams: debug
webrtc: debug
mse: debug
hass: debug
homekit: debug
cameras:
CameraEZVIZC6N:
ffmpeg:
inputs:
- path: rtsp://127.0.0.1:8554/CameraEZVIZC6N # rtsp://USER:[email protected]:554/H.264
input_args: preset-rtsp-restream
roles:
- detect
- record
live:
stream_name: CameraEZVIZC6N
Here is go2rc producer info:
Here is go2rtc consumer info:
Here is go2rtc stream in webRTC mode, I can here the sound:
Here is log when I try to call API go2rtc and send audio to camera:
$ curl --location --request POST 'http://192.168.1.128:1984/api/streams?dst=CameraEZVIZC6N&src=ffmpeg:https://download.samplelib.com/mp3/sample-6s.mp3#audio=opus#input=file'
can't find consumer
$ curl --location --request POST 'http://192.168.1.128:1984/api/streams?dst=CameraEZVIZC6N&src=ffmpeg:https://download.samplelib.com/mp3/sample-6s.mp3#audio=pcma#input=file'
can't find consumer
$ curl --location --request POST 'http://192.168.1.128:1984/api/streams?dst=CameraEZVIZC6N&src=ffmpeg:https://download.samplelib.com/mp3/sample-6s.mp3#audio=pcmu#input=file'
can't find consumer
$ curl --location --request POST 'http://192.168.1.128:1984/api/streams?dst=CameraEZVIZC6N&src=ffmpeg:https://download.samplelib.com/mp3/sample-6s.mp3#audio=aac#input=file'
can't find consumer
Here is the log of go2rtc:
I also tried this external link from go2rtc:
I tried to talk via this microphone but not work:
I also tried to talk via the microphone on my own https link, but not work.
(I already open port 8555 on my router)
Can anyone help me please - I am struggling with the webrtc stream taking a long time to load - loading as MSE first, then after a few seconds switching to RTC.
I need to use the webrtc.create_link service in order to generate a link, which can be used when displaying my Reolink Doorbell as a popup via Pipup on my android tv.
- service: webrtc.create_link
data:
link_id: "{{ link_id }}"
entity: camera.front_doorbell_sub
The only way I can do this is to use the Reolink Integrations camera entity for the sub stream. This stream is awful and takes around 10 seconds or so to actually load, before stuttering and trying freezing. Elsewhere in Home Assistant, I use the go2rtc links to display the camera stream as RTC which works fine, although still takes a couple of seconds to load.
I canāt see any way of using a go2rtc RTC stream in the above script code, as it HAS to use an entity, and go2rtc doesnāt create entities, so I have to use this awful stream source.
Can anyone suggest a way around this? it currently takes 5 or 6 seconds of black screen then loads the MSE stream which displays a static picture for around 4 or 5 seconds, before switching to RTC where it starts displaying fine, but as my popup only stays on the screen for 30 seconds, I barely get any avctual use from it.
Full script:
alias: Display Doorbell PIP Popup on TV
mode: single
variables:
link_id: 0{% for _ in range(39) %}{{ range(10)|random }}{% endfor %}
sequence:
- service: webrtc.create_link
data:
link_id: "{{ link_id }}"
entity: camera.front_doorbell_sub #This comes form the Reolink Integration
open_limit: 1
time_to_live: 120
- service: rest_command.pipup_url_on_tv
data:
ip: 192.168.1.89
duration: 25
title: Front Door
message: Someone is at the front door
width: 640
height: 480
url: >-
https://myhomeassistanturl/webrtc/embed?url={{
link_id }}
Ezviz cameras doesnāt support any open two way audio standard.
Good evening. Everything works fine for me Hikvision KV6113, thanks for the integration. Please tell me there is an LED on the front panel of the Hikvision KV6113, it lights up when the microphone is turned on if you use hik Connect. But if they communicate via a WEB RTC card with two-way audio communication, then the LED lights up and does not go out, that is, the microphone is constantly on, until a reboot. Tell me if there is a command to mute the microphone. Thank you.
ŠŠ±Š°Š²ŃŠµ ŠŗŠ°Š½Š°Š» ISAPI Ń Š¼ŠµŠ½Ń Š½Š° hikvision Š²Š¾Ń ŃŠ°Šŗ.
go2rtc:
streams:
DoorBell:
- rtsp://admin:[email protected]:554/ISAPI/Streaming/Channels/101
- isapi://admin:[email protected]:80/
DoorBellCh2: - rtsp://admin:[email protected]:554/ISAPI/Streaming/Channels/102
- isapi://admin:[email protected]:80/
And you will need HTTPSS:// to have access to two-way audio
You are my hero! After a lot of trying and failing I read your tip and boom I got it! Thank you so much for this input!
Hi is there any way to use this solution with Blink cameras? thx