Go2rtc project - help thread

I wanted to start sound in my recordings through frigate.
Read somewhere that you should update go2rtc has updated to 1.8.5 and now get a lot of errors in my log, read through the documentation that you should fill in according to


"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]" now .

This is how I have tried to fill in my config file.

go2rtc:
  streams:
    #############c310
    carport:
      tapo://[email protected]/stream1
      #- rtsp://min:[email protected]:554/stream1
    carport_sub:
      tapo://[email protected]/stream2
      #- rtsp://min:[email protected]:554/stream2
      #- ffmpeg:carport_sub
   
    ###########C200
    hallen_ny:
       tapo://[email protected]/stream1
     # - rtsp://min:[email protected]:554/stream1
    hallen_ny_sub:
       tapo://[email protected]/stream2
     # - rtsp://min:[email protected]:554/stream2
      #- ffmpeg:hallen_ny_sub



Unsure if this is correct. Should one write one for sub and one for main still? Should you fill in more to get sound in the stream?

Hi
I am trying to access it via Nginx Proxy Manager but I am getting 404: Not Found error. I have added this as custom location on my proxy redirect.

when I go to https://mynpmdomain/go2rtc
What I am doing wrong here ?

Not sure if you figured it out. I have a Tapo C100 working in Frigate using:
go2rtc:
streams:
LivingRoom:
-rtsp://user:[email protected]:554/stream1
- ā€œffmpeg:LivingRoom#audio=aacā€
LivingRoom_sub:
- rtsp://user:[email protected]:554/stream2

sorry the indentation is off but canā€™t figure out how to expand the tiny window they give you to type in. Iā€™m trying to get a new cam - same model - to work but it keeps telling me I have the wrong user and password. It was the same - as the other one - and I even changed it in case it had to be unique to the camera. Still doesnā€™t work but can view in VLC so probably not a camera issue

Hello everyone
I have a Ctronics camera that I would like to integrate.
I have installed the Go2RTC addon and also the RTSPtoWebRTC integration.
I entered my home assistant IP address with port 1984 into the integration.
Then I added the WEBRTC Camera card and entered the URL of my camera there.
When I open the map I have a delay of 10 to 15 seconds until the live image appears.
The GO2RTC logs look like this.

10.3.2024, 17:34:07	info	go2rtc version 1.8.5 linux/amd64
10.3.2024, 17:34:07	info	[rtsp] listen addr=:8554
10.3.2024, 17:34:07	info	[webrtc] listen addr=:8555
10.3.2024, 17:34:07	info	[api] listen addr=:1984
10.3.2024, 17:34:22	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:34:22	debug	[streams] start producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:34:27	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:34:51	debug	[streams] stop producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:35:05	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:35:06	debug	[streams] start producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:35:09	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:36:15	debug	[streams] stop producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:37:00	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:37:00	debug	[streams] start producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:38:10	debug	[streams] stop producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:03	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:03	debug	[streams] start producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:08	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:14	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:24	debug	[streams] stop producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:28	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:28	debug	[streams] start producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:34	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:42:45	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:43:02	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:554/12
10.3.2024, 17:43:02	debug	[streams] stop producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:43:40	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:43:40	debug	[streams] start producer url=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:43:53	debug	[webrtc] new consumer src=rtsp://admin:[email protected]:8554/12
10.3.2024, 17:44:09	debug	[streams] stop producer 

Where is the error or is the delay normal?

Thank you in advance for the help
Volker

WebRTC card canā€™t be ā€œopenedā€. Are you sure you using it and not default card?

Normal delay for almost any stream is sub 0.5 second.

Hi all. I apologize if this has been answered many times already. Gets hard to find answers as these threads get larger!

I am using the go2rtc addon to bring an aqara cam in using the HK device integration of HA.

Separately I am running Frigate with some Amcrest cams as well as pulling in the aqara stream. In my Frigate config I am not currently running the Amcrest cams through go2rtc. Should I? What is the added benefit of setting it up this way?

Thanks.

Hello
Thanks for your help . Iā€™m actually sure Iā€™m using the right card. Here is the code.

type: custom:webrtc-camera
url: rtsp://admin:[email protected]:8554/11
mode: webrtc

Do you have another tip?
Thanks

So I have now changed my configuration again. I have inserted my camera into the go2rtc.yaml.

streams:
    haustuer: rtsp://nutzer:[email protected]:8554/

I have now set the webRTC card like this.

type: custom:webrtc-camera
url: haustuer
mode: webrtc

But it still doesnā€™t work. When I open the map, I always have a delay of over 10 seconds until the image appears.
Until then it says loding at the top right.
What have I done wrong? Please help.
Thanks

Well. Do you have start delay and not real time delay?
Start delay is a normal thing for some cameras.
For good cameras, with little key frame interval, it also less than second.

Yes, you are right, the delay only occurs at startup. The running stream is smooth and time synchronous. If I enter the rtsp address of my camera into the VLC player, it takes about 2 seconds and then the image appears. If the error was in the camera there would also have to be a delay when starting.
Do you have another idea?

Hello
I think Iā€™ve found the problem.
If it says in the go2rtc logs

[streams] start producer url=rtsp://nutzer:[email protected]:8554/11

Then it takes up to 15 seconds for the image to appear.
but the server is stopped
then the picture comes immediately.

[streams] stop producer url=rtsp://nutzer:[email protected]:8554/11

I have integrated a small live image from the camera into my standard dashboard. When the bell rings, it switches to the camera dashboard with the large image of the camera.
For example, do I change From the energy board to the camera, the image is there immediately.
Is there a solution for this?
Thanks

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.

Intermittent sound and voice on two-way audio RTSP, ISAPI, Hikvision DS-KV6113, GO2RTC, WEBRTC, WEBRTC-CAMERA, media_player: - platform: webrtc

Good afternoon. Please help.
I have a Hikvision DS-KV6113 doorbell added to Home Assistant via Frigate go2rtc in this way:
go2rtc:
streams:
DoorBell:

I added media players to the configuration.yaml file like this:
media_player:

  • platform: webrtc
    name: PlayerDoorBell
    stream: DoorBell
    audio: pcmu/8000
  • platform: webrtc
    name: PlayerDoorBellCh2
    stream: DoorBellCh2
    audio: pcmu/8000

In the web interface I added a camera via the webrtc-camera video and sound card:
type: custom:webrtc-camera
url: DoorBellCh2
mode: webrtc
media: video, audio
muted: true
background: true
style: ā€˜video {aspect-ratio: 16/9; object-fit: fill;}ā€™

And the camera via the webrtc-camera card video, sound and microphone:
type: custom:webrtc-camera
url: DoorBellCh2
mode: webrtc
media: video, audio, microphone
muted: true
background: true
style: ā€˜video {aspect-ratio: 16/9; object-fit: fill;}ā€™

So, if you make multiple connections to a two-way audio camera, or while watching a two-way audio camera, play some audio file through the media player, the sound on that camera will become quiet and choppy, and if you speak into the microphone, the voice will become choppy. I tried changing different firmware on the camera, but it didnā€™t help. If you reboot the frigate go2rtc everything becomes normal and works for some time.
I also noticed that when this happens, the microphone on the camera and on the browser tab works for a long time, and the Android phone can generally transmit sound and voice even after closing and unloading the application.
Is it possible to somehow fix this or add some command to reload the stream very quickly. Help me please. Thank you.

@AlexxIT
i want to add the Netatmo Presence stream

http://192.168.1.220/adb9045c5d44c94aqwwedsa4f1b164/live/index.m3u8

when i out it in to the config like this

streams:
    test:
      - http://192.168.1.220/adb9045c5d44cavseed83fc6b4f1b164/live/index.m3u8#video=h264

i get a picture but it is extrem laggy

do you have a solution to get the video smooth?

FFmpeg source best solution when built-in go2rtc sources has problems.

i think i walked a bit i the right direction. but now i need help. i used ffmpeg and rtsp server in docker and i have now a working stream.

  ffmpeg_anhanger_vorn:
    image: lscr.io/linuxserver/ffmpeg:latest
    container_name: ffmpeg_anhanger_vorn
    command: >
      -re -i http://192.168.1.174/372123456c7a59de72da57c27bc76e5e/live/index.m3u8
      -c:v copy -c:a aac -ar 44100 -b:a 128k
      -f rtsp rtsp://rtsp-simple-server:8554/anhanger_vorn
    depends_on:
      - rtsp-simple-server

how do i get the ffmpeg parameter into the go2rtc config? i tried this but it is not working:

streams:
  anhanger_vorn:
    - input: ffmpeg http://192.168.1.174/312345670ac7a59de72da57c27bc76e5e/live/index.m3u8 -c:v copy -c:a aac -ar 44100 -b:a 128k

???

streams:
  anhanger_vorn:
    - ffmpeg:http://192.168.1.174/312345670ac7a59de72da57c27bc76e5e/live/index.m3u8#video=copy#audio=aac

Are you sure you needs audio transcoding? Usually HLS has AAC codec. You can use it as is.

some magic for netatmo cameras:

streams:
  yard:
    - ffmpeg:http://192.168.88.158/*****/live/files/high/index.m3u8#audio=opus/16000#hardware#video=h264#input=netatmo_input`

ffmpeg:
    netatmo_input: "-err_detect ignore_err -y -fflags +genpts -thread_queue_size 512 -analyzeduration 20000000 -re -i {input}"
1 Like

I hope youā€™re all doing well. Iā€™m currently exploring the Go2RTC Hardware Acceleration feature in Home Assistant and have some questions about it. Specifically, Iā€™ve noticed that enabling hardware acceleration doesnā€™t seem to provide clear advantages with my general RTSP streams.

Could someone clarify whether the stream is actually rendered on the host after Hardware Acceleration is activated, or does it happen later at the client? Iā€™m not entirely clear on how this process works and whether there are specific scenarios where Hardware Acceleration does offer noticeable benefits.