Mirrors/blakeblackshear.frigate
1
0
Fork 0
mirror of https://github.com/blakeblackshear/frigate.git synced 2025-10-27 10:52:11 +01:00
Code Issues Projects Releases Wiki Activity

clarify webrtc for two way talk (#19683)

Browse Source
This commit is contained in:
Josh Hawkins 2025-08-21 05:43:07 -05:00 committed by GitHub
parent 664a6fd0cb
commit 8a01643acf
No known key found for this signature in database
GPG Key ID: B5690EEEBB952194
2 changed files with 2 additions and 2 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
docs/docs/frigate/installation.md
View File

@ -43,7 +43,7 @@ The following ports are used by Frigate and can be mapped via docker as required
| `8971` | Authenticated UI and API access without TLS. Reverse proxies should use this port. |
| `5000` | Internal unauthenticated UI and API access. Access to this port should be limited. Intended to be used within the docker network for services that integrate with Frigate. |
| `8554` | RTSP restreaming. By default, these streams are unauthenticated. Authentication can be configured in go2rtc section of config. |
| `8555` | WebRTC connections for low latency live views. |
| `8555` | WebRTC connections for cameras with two-way talk support. |
#### Common Docker Compose storage configurations

2
docs/docs/guides/configuring_go2rtc.md
View File

@ -114,7 +114,7 @@ section.
## Next steps
1. If the stream you added to go2rtc is also used by Frigate for the `record` or `detect` role, you can migrate your config to pull from the RTSP restream to reduce the number of connections to your camera as shown [here](/configuration/restream#reduce-connections-to-camera).
2. You may also prefer to [setup WebRTC](/configuration/live#webrtc-extra-configuration) for slightly lower latency than MSE. Note that WebRTC only supports h264 and specific audio formats and may require opening ports on your router.
2. You can [set up WebRTC](/configuration/live#webrtc-extra-configuration) if your camera supports two-way talk. Note that WebRTC only supports h264 and specific audio formats and may require opening ports on your router.
## Important considerations