Go2rtc project - help thread

Third party integrations
41

Oh, i removed the slashes after rtsp (typo), sorry.
Have i forgotten anything else? Checked the example in the docu, but cant find my fault.

Thanks

42

Everything should work now

43

After a few seconds i receive a very laggy stream. Audio and video are very jerky. Also tried to add pcmu for audio, but no difference.

44

Likely the hardware you’re doing transcoding on (running go2rtc) is making it jerky.

45

Is there a way to adjust the timeout (increase) for the ‘Timeout Handling WebRTC offer’? I have a couple cameras that run through ffmpeg with go2rtc and when loading at once sometimes some fail to display. This is likely due to hardware slowness but rather than ‘Failed to start WebRTC’, i’d prefer to either a) increase the timeout or b) retry itself.

Once it fails, I’d need to refresh the entire lovelace for it to retry again


46

Its running as home Assistant Plugin, the ha vm has two cores and 4gig of RAM. When i Start the stream the cpu is at about 70% and RAM at 40%. Do you have suggestions for improvement?

47

Transcoding video is not chip operation. Hardware acceleration is very complicated task for all possible user setups. Native H265 support is very limited for all possible user devices.

Some of this problems will be solved in future go2rtc updates.

1 Like
48

@AlexxIT - my Dahua doorbell camera has never worked with any of your plugins. With go2RTC this is the error I’m getting. Any suggestions on how I can resolve? I am able to see the RTSP stream in VLC totally fine with the same URL.

07:32:41.702 INF [rtsp] listen addr=:8554
07:32:41.705 INF [hass] load stream url=hass:Camera_2_h264
07:32:41.705 INF [hass] load stream url=hass:Door_Camera
07:32:41.705 INF [hass] load stream url=hass:192_168_0_69
07:32:41.705 INF [api] listen addr=:1984
07:32:41.705 INF [srtp] listen addr=:8443
2022/09/07 13:03:04 [INFO] mdns: Closing client {true false 0xc00012c770 <nil> 0xc00012c778 <nil> 1 0xc000100780}
07:33:50.180 INF [streams] create new stream url=rtsp://admin:[email protected]:554/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
07:33:53.017 WRN [api.webrtc] add consumer error="couldn't find the matching tracks"
49

@sachinss increase log level to trace and came to github issues. Or you can find me in Telegram with same nick.

50

@calisro next beta.4 will have really fast start for streams with ffmpeg source only for audio transcoding. Something like this:

streams:
  dahua_opus:
    - rtsp://192.168.1.123/cam/realmonitor?channel=1&subtype=0
    - ffmpeg:rtsp://192.168.1.123/cam/realmonitor?channel=1&subtype=0#audio=opus

It’s not recommended to use #video=copy#audio=something if it possible. Because ffmpeg will start more than 3 seconds in this case. But stream delay will be OK anyway (less than half second as usual). Better to split stream on two sources (direct rtsp and ffmpeg).

1 Like
51

Thanks for confirming. I am using this already for 4 streams.

  backyard_main:
    - hass:10_100_1_134
    - ffmpeg:rtsp://127.0.0.1:8554/hass%3A10_100_1_134#audio=opus

It works good except when loading many cameras (6) at once. Then I always get 1 or 2 that timeout with ‘Timeout Handling WebRTC offer’. I’ll try beta4 when its out but i’d like to avoid cameras not loading. I’d rather wait a little longer for them to load.

53

I have go2rtc and RTSPtoWebRTC installed. I see my camera picked up by the RTSPtoWebRTC logs but go2rtc doesn’t seem to see it. It seems like the only way for the stream to be WebRTC is if I manually add my rtsp link in the go2rtc.yaml but then I can click on the webrtc link in the UI of go2rtc but in my dashboard the stream still isn’t webrtc.

54

Can you explain what you mean here? I don’t think there are logs specific to that integration itself. So trying to understand. When you added ‘RTSPtoWebRTC’ did you add the api? (probably http://127.0.0.1:1984/). Did you remove the other ‘webrtc camera’ integration if you had it installed (it would interfere)?

That isn’t required but you could


How are you checking that?

Anyway, you can define whatever streams you want in the go2rtc too and then reference those as rtsp streams in frigate or generic camera using its rtsp address. (rtsp://127.0.0.1:8554/[rtsp2rtc config name])

But it should be zero config in that it converts them to webrtc.

Another way to see if it is is to look at the attributes of your camera after/while you view it. You’ll see an attribute like this:

“frontend_stream_type: web_rtc”

image
image944×114 13.9 KB

55

I see this in the RTSPtoWebRTC logs [GIN] 2022/09/07 - 15:29:31 | 500 | 5.063519892s | 10.10.0.12 | POST "/stream" [GIN] 2022/09/07 - 15:33:57 | 404 | 3.93”s | 10.10.0.12 | GET "/streams" [GIN] 2022/09/07 - 15:33:57 | 200 | 160.84”s | 10.10.0.12 | GET "/static/"

I don’t see any configuration for RTSPtoWebRTC so I’m not sure where to add the API.

I’ll check the attributes of my camera but when I was using WebRTC before the video had 0 lag and was super smooth, but currently it’s very delayed and choppy.

56

Remove it and re-add it. It should look like this. I had to add that when I initially installed it.

image

1 Like
57

Thank you for that. I added that, but I’m still running into the same issues. go2rtc automatically added my rtmp camera stream but not rtsp. I think it has to do with the way frigate is sending the stream over. When I go through RTMP it shows

“frontend_stream_type: web_rtc”

but there is a huge delay because of RTMP from Frigate

But if I disable RTMP in Frigate it doesn’t work at all.

I do see these messages in go2rtc

23:06:40.622 WRN [api.webrtc] add consumer error="couldn't find the matching tracks"
23:06:40.623 ERR [api.hass] exchange SDP error="couldn't find the matching tracks"
23:06:45.202 WRN [api.webrtc] add consumer error="couldn't find the matching tracks"
23:06:45.204 ERR [api.hass] exchange SDP error="couldn't find the matching tracks"
58

I don’t know frigate. Maybe someone else can chime in. But my guess is you still want the streams coming into frigate the same way but then you want the UI to use webrtc for your viewing. If that is the case, i’d think you’d add the rtsp url of a stream into go2rtc.yaml and then potentially create generic cameras to those entries and then you can add those new generic cameras to your UI. I am really just thinking out loud as I haven’t used frigate.

59

Thanks for that idea! I was hoping not to have double camera entries but if I don’t have a choice I’ll go down that route. Thanks again for all of your help.

60

if you camera supports webrtc stream, you can use native HA streaming functionality as it supports webrtc streaming now.

if your camera is streaming in rtsp, you have 3 options ahead;

  • use rtsptowebrtc add-on and integration with native HA streaming. Add-on will convert the stream into webrtc on the fly.
  • use webrtc integration, it will execute a binary in the server and convert rtsp stream into webrtc. This will use a custom player on frontend. Because of some codec issues we are facing, next solution comes into place.
  • use go2rtc add-on, similar to first add-on, this should convert your rtsp streams into webrtc on the fly with better codec support.

I am still going through details of all these to understand better.

61

Yes, you right. But go2rtc is not just about RTSP.
It can support RTSP, RTMP, HTTP, MJPEG, HLS, USB Cameras, HomeKit Cameras, files, etc.
Also it can convert codecs when it needed and do other nice things with ffmpeg.
Also it support 2-way audio.
Also it can stream to RTSP, WebRTC, MSE, mp4 (in development), HomeKit (in development)

Also it has better support for WebRTC with external access.
And many other things in the future
 :slight_smile: