Thanks for the suggestion. Sorry, but I clearly am missing a piece of the puzzle.
I’m still randomly getting static images of cameras on the frigate cards and sometimes live video with very little latency. The WebRTC card is just a black card with a pause logo and the disconnected icon in top right.
I currently have WebRTCCamera and RTSPtoWebRTC integrations. The former has min and max UDP port set to 0 (default AFAIK). The latter has the url http://127.0.0.1:1984 and has google STUN set as
I’ve created a test page with a Frigate card, which still works occasionally and two WebRTC Camera cards with the following config
Here are the logs from Go2RTC. It appears that it’s setting up the streams (mostly from the frigate cards I think as it has all cameras rather than just the 2 I have on the test page). It still lists a couple of IO timeouts as mentioned previously.
You don’t neet RTSPtoWebRTC integrations and set any STUN when you using new WebRTC Camera v3. It support multiple technologies for get stream from go2rtc. If you will have problem with external access with WebRTC - it will use MSE. Also it support MJPEG and MP4 (not tested yet).
The problem is only if you use an iPhone. Because it does not support MSE. And you should setup external access with WebRTC correctly. It can be difficult if you don’t understand much about IT…
Thanks again. Yes, as I mentioned above I am using an iPhone and understand it has limitations but was also under the impression it should be falling back to HLS?
I’ll make the changes above and report back. Thanks again for the assistance and a great add on as it’s working amazing internally for me!
I would first remove the RTSP integration and then download the beta version of the WebRTC Camera from HACS. the public release version of WebRTC camera doesn’t yet support Go2RTC. Then from there I’d test, I haven’t been successful myself in getting either Frigate card or WebRTC card to reliable work when I am not on my LAN network.
As for the latest WebRTC card beta it doesn’t even work on LAN, so I recommend using one release older beta version if you experience issues on your LAN network too.
Seems like the latest version of the Go2RTC addon is giving me issues, its slow to load and sometimes doesn’t load at all. I have to restart the addon for the streams to load.
Hey @AlexxIT , I am having troubles with the RC7 for the addon and Beta 4 for WebRTC Camera. After a few hours of not viewing any camera streams the streams never start. It just gets stuck on Loading2 and never starts, the only way to get the streams started is to restart the addon. After that the streams will work for a few hours then do the same thing again. Interestingly if I use Frigate card the streams start albeit a little slowly, so not sure if this is an integration or addon issues. I am not seeing any errors or anything in the logs either.
Amazing! It seems it’s now all finally working. Thanks @pabla and @AlexxIT
I basically started from scratch and removed WebRTCCamera, RTSPtoWebRTC and Go2RTC.
I then redownloaded WebRTC Camera v3 beta 5 and Go2RTC 0.1rc8 and restarted HA. I then recreated my camera cards using the WebRTCv3 cards and the camera entitties. Unfortunately I was getting an error “webrtc the type provided ('video/mp4; codecs=“undefined”) not supported” on WebRTC Camera v3 beta 5. After rolling back to beta 4 the error left and my cameras were all working using RTC on LAN via PC and iPhone and MP4 on iPhone when external.
The card via Mac/PC Edge browser allows me to turn the audio off, however the iPhone companion app doesn’t seem to allow this. Any chance of adding this function? With 10 cameras all playing audio it’s a bit much, but it’s amazing to finally have audio via HA
Happy to hear it worked out for you! I’m still struggling a bit myself but seeing your success has made me confident that I will be able to figure it out!
As for the audio what I noticed is when using MP4 stream on iPhone the native iOS video controls are hidden. You can permanently mute each stream by adding ‘muted: true’ in each camera card config but I’m not sure if you this is what you want.
yeah, ideally it would be good to default to mute but re-enable on demand, but that will suffice for now. Thanks again for the suggestion and good luck with your setup.
Edit: actually, the muted: true works great. It mutes by default but you can click to unmute. Thanks again
Now that i’ve got this working on my setup, i’m attempting to setup another environment I look after. All works fine via browser although it’s using MSE. Go2RTC is reporting an error which I believe is due to the password on the cameras:
parse "rtsp:admin:xxxx": invalid port ":xxx" after host
The password has a # followed by some numbers. xxx is the first part of the password prior to the hash
So my setup is currently struggling to start a live stream after the addon has been started for some time. Basically after I start the addon, streams start as expected. After maybe about 20-30 mins they stop starting. Taking a look at the go2rtc webpage and the streams have disappeared and here is what the logs look like. I have tried a chrome browser and the iPhone app both do not get a live stream. While this is happening, if I use Frigate card instead of WebRTC camera card, the stream starts but it looks like it falls back to HLS since the stream is very delayed and I don’t see anything in the logs for the go2RTC addon about a new consumer.
I am running go2rtc addon RC9 and WebRTC Beta 5.
Edit: Just as a test I added the streams to the go2rtc config just to see if that would help and after a few minutes they disappeared from the webUI. Seems like a possible memory leak issue
20:20:37.434 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=1&subtype=0
20:20:37.443 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=6&subtype=0
20:20:37.443 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=2&subtype=0
20:20:37.546 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=5&subtype=0
20:20:37.558 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=12&subtype=0
20:20:37.559 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=8&subtype=0
20:20:37.562 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=3&subtype=0
20:20:38.010 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=7&subtype=0
20:20:38.011 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=4&subtype=0
20:20:38.016 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=10&subtype=0
20:20:38.049 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=9&subtype=0
20:20:38.049 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=11&subtype=0
20:21:18.658 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=1&subtype=0
20:21:18.670 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=2&subtype=0
20:21:18.699 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=6&subtype=0
20:21:18.726 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=5&subtype=0
20:21:18.758 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=12&subtype=0
20:21:18.776 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=8&subtype=0
20:21:18.811 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=7&subtype=0
20:21:18.823 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=3&subtype=0
20:21:18.839 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=4&subtype=0
20:21:18.862 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=10&subtype=0
20:21:18.882 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=11&subtype=0
20:21:18.902 DBG [webrtc] new consumer url=rtsp://xxxxx:[email protected]:554/cam/realmonitor?channel=9&subtype=0
I have a strange problem. I use go2rtc in docker container with own turn server and rstp2webrtc integration in hasio. If i connect from local network everything works correctly, if i connect from lte network stream doesn’t work. Port 8555 is forwarded both udp and tcp.
If i install go2rtc as a hasio addon with same config, external streaming works perfectly.
LE: Found the problem. I had two dst-nat rules on router.
Hey guys , I have been banging my head with iframe and browsermod-popup for days now .
I have setup the NPM and Cloudflared to work and I can access my camera externally with 2 way audio .
I can access my camera using https://go2.mydomain/webrtc.html?src=doorbell_cam when I visit this webpage I can see the Mic icon on webpage and in my ios orange icon showing that mic is being used and also 2 way audio works with direct links.
But when I use in picture-elements card iframe only shows the stream 2 way audio is not working
I have foscam FI9000EP camera.
My config: in docker go2rtc and 3.01 webrtc in home assistant configuration
This is my rtps address “rtsp://user:[email protected]:88/videoMain” which is working on go2rtc page 192.168.1.222:1984
Set it up on my HA dashboard with this:
I see the live video and the RTC logo on the right corner. But the live video always stutters.
If I click full screen the picture is the same stutters.
On go2rtc page when I click stream get same stutters but if click mp4 it is open full screen and perfect the live video without stutters.
Want to make a new generic cam with this setup but get this error and not working with mp4:
In last few days telegram notification that get pushed from go2rtc are not working anymore.
Don’t know if due to latest version of go2rtc or latest supervisor or Hass update…