diff --git a/docker/main/Dockerfile b/docker/main/Dockerfile index 8625a63a8..91279115f 100644 --- a/docker/main/Dockerfile +++ b/docker/main/Dockerfile @@ -33,7 +33,7 @@ RUN --mount=type=tmpfs,target=/tmp --mount=type=tmpfs,target=/var/cache/apt \ FROM scratch AS go2rtc ARG TARGETARCH WORKDIR /rootfs/usr/local/go2rtc/bin -ADD --link --chmod=755 "https://github.com/AlexxIT/go2rtc/releases/download/v1.6.2/go2rtc_linux_${TARGETARCH}" go2rtc +ADD --link --chmod=755 "https://github.com/AlexxIT/go2rtc/releases/download/v1.7.1/go2rtc_linux_${TARGETARCH}" go2rtc #### diff --git a/docker/main/rootfs/usr/local/go2rtc/create_config.py b/docker/main/rootfs/usr/local/go2rtc/create_config.py index aefed5f83..7d69dc415 100644 --- a/docker/main/rootfs/usr/local/go2rtc/create_config.py +++ b/docker/main/rootfs/usr/local/go2rtc/create_config.py @@ -100,12 +100,25 @@ for name in go2rtc_config.get("streams", {}): stream = go2rtc_config["streams"][name] if isinstance(stream, str): - go2rtc_config["streams"][name] = go2rtc_config["streams"][name].format( - **FRIGATE_ENV_VARS - ) + try: + go2rtc_config["streams"][name] = go2rtc_config["streams"][name].format( + **FRIGATE_ENV_VARS + ) + except KeyError as e: + print( + "[ERROR] Invalid substitution found, see https://docs.frigate.video/configuration/restream#advanced-restream-configurations for more info." + ) + sys.exit(e) + elif isinstance(stream, list): for i, stream in enumerate(stream): - go2rtc_config["streams"][name][i] = stream.format(**FRIGATE_ENV_VARS) + try: + go2rtc_config["streams"][name][i] = stream.format(**FRIGATE_ENV_VARS) + except KeyError as e: + print( + "[ERROR] Invalid substitution found, see https://docs.frigate.video/configuration/restream#advanced-restream-configurations for more info." + ) + sys.exit(e) # add birdseye restream stream if enabled if config.get("birdseye", {}).get("restream", False): diff --git a/docs/docs/configuration/advanced.md b/docs/docs/configuration/advanced.md index bc1a7c421..e32795946 100644 --- a/docs/docs/configuration/advanced.md +++ b/docs/docs/configuration/advanced.md @@ -120,7 +120,7 @@ NOTE: The folder that is mapped from the host needs to be the folder that contai ## Custom go2rtc version -Frigate currently includes go2rtc v1.6.2, there may be certain cases where you want to run a different version of go2rtc. +Frigate currently includes go2rtc v1.7.1, there may be certain cases where you want to run a different version of go2rtc. To do this: diff --git a/docs/docs/configuration/camera_specific.md b/docs/docs/configuration/camera_specific.md index bbfb03778..f092ec325 100644 --- a/docs/docs/configuration/camera_specific.md +++ b/docs/docs/configuration/camera_specific.md @@ -80,8 +80,8 @@ cameras: rtmp: enabled: False # <-- RTMP should be disabled if your stream is not H264 detect: - width: # <- optional, by default Frigate tries to automatically detect resolution - height: # <- optional, by default Frigate tries to automatically detect resolution + width: # <- optional, by default Frigate tries to automatically detect resolution + height: # <- optional, by default Frigate tries to automatically detect resolution ``` ### Blue Iris RTSP Cameras @@ -108,20 +108,20 @@ According to [this discussion](https://github.com/blakeblackshear/frigate/issues ```yaml go2rtc: streams: - your_reolink_camera: + your_reolink_camera: - "ffmpeg:http://reolink_ip/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=username&password=password#video=copy#audio=copy#audio=opus" - your_reolink_camera_sub: + your_reolink_camera_sub: - "ffmpeg:http://reolink_ip/flv?port=1935&app=bcs&stream=channel0_ext.bcs&user=username&password=password" cameras: - reolink: + your_reolink_camera: ffmpeg: inputs: - - path: rtsp://127.0.0.1:8554/your_reolink_camera?video=copy&audio=aac + - path: rtsp://127.0.0.1:8554/your_reolink_camera input_args: preset-rtsp-restream roles: - record - - path: rtsp://127.0.0.1:8554/your_reolink_camera_sub?video=copy + - path: rtsp://127.0.0.1:8554/your_reolink_camera_sub input_args: preset-rtsp-restream roles: - detect @@ -140,7 +140,7 @@ go2rtc: - rtspx://192.168.1.1:7441/abcdefghijk ``` -[See the go2rtc docs for more information](https://github.com/AlexxIT/go2rtc/tree/v1.6.2#source-rtsp) +[See the go2rtc docs for more information](https://github.com/AlexxIT/go2rtc/tree/v1.7.1#source-rtsp) In the Unifi 2.0 update Unifi Protect Cameras had a change in audio sample rate which causes issues for ffmpeg. The input rate needs to be set for record and rtmp if used directly with unifi protect. diff --git a/docs/docs/configuration/index.md b/docs/docs/configuration/index.md index 874a77cda..c7db4250d 100644 --- a/docs/docs/configuration/index.md +++ b/docs/docs/configuration/index.md @@ -436,7 +436,7 @@ rtmp: enabled: False # Optional: Restream configuration -# Uses https://github.com/AlexxIT/go2rtc (v1.6.2) +# Uses https://github.com/AlexxIT/go2rtc (v1.7.1) go2rtc: # Optional: jsmpeg stream configuration for WebUI diff --git a/docs/docs/configuration/live.md b/docs/docs/configuration/live.md index 63d83e4d9..9ffd4114b 100644 --- a/docs/docs/configuration/live.md +++ b/docs/docs/configuration/live.md @@ -78,7 +78,7 @@ WebRTC works by creating a TCP or UDP connection on port `8555`. However, it req - 192.168.1.10:8555 - stun:8555 ``` - + - For access through Tailscale, the Frigate system's Tailscale IP must be added as a WebRTC candidate. Tailscale IPs all start with `100.`, and are reserved within the `100.0.0.0/8` CIDR block. :::tip @@ -115,4 +115,4 @@ services: ::: -See [go2rtc WebRTC docs](https://github.com/AlexxIT/go2rtc/tree/v1.6.2#module-webrtc) for more information about this. +See [go2rtc WebRTC docs](https://github.com/AlexxIT/go2rtc/tree/v1.7.1#module-webrtc) for more information about this. diff --git a/docs/docs/configuration/restream.md b/docs/docs/configuration/restream.md index 7f4c73717..405ffa7d4 100644 --- a/docs/docs/configuration/restream.md +++ b/docs/docs/configuration/restream.md @@ -7,7 +7,7 @@ title: Restream Frigate can restream your video feed as an RTSP feed for other applications such as Home Assistant to utilize it at `rtsp://:8554/`. Port 8554 must be open. [This allows you to use a video feed for detection in Frigate and Home Assistant live view at the same time without having to make two separate connections to the camera](#reduce-connections-to-camera). The video feed is copied from the original video feed directly to avoid re-encoding. This feed does not include any annotation by Frigate. -Frigate uses [go2rtc](https://github.com/AlexxIT/go2rtc/tree/v1.6.2) to provide its restream and MSE/WebRTC capabilities. The go2rtc config is hosted at the `go2rtc` in the config, see [go2rtc docs](https://github.com/AlexxIT/go2rtc/tree/v1.6.2#configuration) for more advanced configurations and features. +Frigate uses [go2rtc](https://github.com/AlexxIT/go2rtc/tree/v1.7.1) to provide its restream and MSE/WebRTC capabilities. The go2rtc config is hosted at the `go2rtc` in the config, see [go2rtc docs](https://github.com/AlexxIT/go2rtc/tree/v1.7.1#configuration) for more advanced configurations and features. :::note @@ -53,31 +53,31 @@ One connection is made to the camera. One for the restream, `detect` and `record ```yaml go2rtc: streams: - rtsp_cam: # <- for RTSP streams + name_your_rtsp_cam: # <- for RTSP streams - rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio - - "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus) - http_cam: # <- for other streams + - "ffmpeg:name_your_rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus) + name_your_http_cam: # <- for other streams - http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=user&password=password # <- stream which supports video & aac audio - - "ffmpeg:http_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus) + - "ffmpeg:name_your_http_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus) cameras: - rtsp_cam: + name_your_rtsp_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - - path: rtsp://127.0.0.1:8554/rtsp_cam # <--- the name here must match the name of the camera in restream + - path: rtsp://127.0.0.1:8554/name_your_rtsp_cam # <--- the name here must match the name of the camera in restream input_args: preset-rtsp-restream roles: - record - detect - audio # <- only necessary if audio detection is enabled - http_cam: + name_your_http_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - - path: rtsp://127.0.0.1:8554/http_cam # <--- the name here must match the name of the camera in restream + - path: rtsp://127.0.0.1:8554/name_your_http_cam # <--- the name here must match the name of the camera in restream input_args: preset-rtsp-restream roles: - record @@ -92,44 +92,44 @@ Two connections are made to the camera. One for the sub stream, one for the rest ```yaml go2rtc: streams: - rtsp_cam: + name_your_rtsp_cam: - rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg - - "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus - rtsp_cam_sub: + - "ffmpeg:name_your_rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus + name_your_rtsp_cam_sub: - rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg - - "ffmpeg:rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus - http_cam: + - "ffmpeg:name_your_rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus + name_your_http_cam: - http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=user&password=password # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg - - "ffmpeg:http_cam#audio=opus" # <- copy of the stream which transcodes audio to opus - http_cam_sub: + - "ffmpeg:name_your_http_cam#audio=opus" # <- copy of the stream which transcodes audio to opus + name_your_http_cam_sub: - http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_ext.bcs&user=user&password=password # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg - - "ffmpeg:http_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus + - "ffmpeg:name_your_http_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus cameras: - rtsp_cam: + name_your_rtsp_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - - path: rtsp://127.0.0.1:8554/rtsp_cam # <--- the name here must match the name of the camera in restream + - path: rtsp://127.0.0.1:8554/name_your_rtsp_cam # <--- the name here must match the name of the camera in restream input_args: preset-rtsp-restream roles: - record - - path: rtsp://127.0.0.1:8554/rtsp_cam_sub # <--- the name here must match the name of the camera_sub in restream + - path: rtsp://127.0.0.1:8554/name_your_rtsp_cam_sub # <--- the name here must match the name of the camera_sub in restream input_args: preset-rtsp-restream roles: - audio # <- only necessary if audio detection is enabled - detect - http_cam: + name_your_http_cam: ffmpeg: output_args: record: preset-record-generic-audio-copy inputs: - - path: rtsp://127.0.0.1:8554/http_cam # <--- the name here must match the name of the camera in restream + - path: rtsp://127.0.0.1:8554/name_your_http_cam # <--- the name here must match the name of the camera in restream input_args: preset-rtsp-restream roles: - record - - path: rtsp://127.0.0.1:8554/http_cam_sub # <--- the name here must match the name of the camera_sub in restream + - path: rtsp://127.0.0.1:8554/name_your_http_cam_sub # <--- the name here must match the name of the camera_sub in restream input_args: preset-rtsp-restream roles: - audio # <- only necessary if audio detection is enabled @@ -138,7 +138,7 @@ cameras: ## Advanced Restream Configurations -The [exec](https://github.com/AlexxIT/go2rtc/tree/v1.6.2#source-exec) source in go2rtc can be used for custom ffmpeg commands. An example is below: +The [exec](https://github.com/AlexxIT/go2rtc/tree/v1.7.1#source-exec) source in go2rtc can be used for custom ffmpeg commands. An example is below: NOTE: The output will need to be passed with two curly braces `{{output}}` diff --git a/docs/docs/guides/configuring_go2rtc.md b/docs/docs/guides/configuring_go2rtc.md index d242086d0..8c9a492af 100644 --- a/docs/docs/guides/configuring_go2rtc.md +++ b/docs/docs/guides/configuring_go2rtc.md @@ -11,7 +11,7 @@ Use of the bundled go2rtc is optional. You can still configure FFmpeg to connect # Setup a go2rtc stream -First, you will want to configure go2rtc to connect to your camera stream by adding the stream you want to use for live view in your Frigate config file. If you set the stream name under go2rtc to match the name of your camera, it will automatically be mapped and you will get additional live view options for the camera. Avoid changing any other parts of your config at this step. Note that go2rtc supports [many different stream types](https://github.com/AlexxIT/go2rtc/tree/v1.6.2#module-streams), not just rtsp. +First, you will want to configure go2rtc to connect to your camera stream by adding the stream you want to use for live view in your Frigate config file. If you set the stream name under go2rtc to match the name of your camera, it will automatically be mapped and you will get additional live view options for the camera. Avoid changing any other parts of your config at this step. Note that go2rtc supports [many different stream types](https://github.com/AlexxIT/go2rtc/tree/v1.7.1#module-streams), not just rtsp. ```yaml go2rtc: @@ -24,7 +24,7 @@ The easiest live view to get working is MSE. After adding this to the config, re ### What if my video doesn't play? -If you are unable to see your video feed, first check the go2rtc logs in the Frigate UI under Logs in the sidebar. If go2rtc is having difficulty connecting to your camera, you should see some error messages in the log. If you do not see any errors, then the video codec of the stream may not be supported in your browser. If your camera stream is set to H265, try switching to H264. You can see more information about [video codec compatibility](https://github.com/AlexxIT/go2rtc/tree/v1.6.2#codecs-madness) in the go2rtc documentation. If you are not able to switch your camera settings from H265 to H264 or your stream is a different format such as MJPEG, you can use go2rtc to re-encode the video using the [FFmpeg parameters](https://github.com/AlexxIT/go2rtc/tree/v1.6.2#source-ffmpeg). It supports rotating and resizing video feeds and hardware acceleration. Keep in mind that transcoding video from one format to another is a resource intensive task and you may be better off using the built-in jsmpeg view. Here is an example of a config that will re-encode the stream to H264 without hardware acceleration: +If you are unable to see your video feed, first check the go2rtc logs in the Frigate UI under Logs in the sidebar. If go2rtc is having difficulty connecting to your camera, you should see some error messages in the log. If you do not see any errors, then the video codec of the stream may not be supported in your browser. If your camera stream is set to H265, try switching to H264. You can see more information about [video codec compatibility](https://github.com/AlexxIT/go2rtc/tree/v1.7.1#codecs-madness) in the go2rtc documentation. If you are not able to switch your camera settings from H265 to H264 or your stream is a different format such as MJPEG, you can use go2rtc to re-encode the video using the [FFmpeg parameters](https://github.com/AlexxIT/go2rtc/tree/v1.7.1#source-ffmpeg). It supports rotating and resizing video feeds and hardware acceleration. Keep in mind that transcoding video from one format to another is a resource intensive task and you may be better off using the built-in jsmpeg view. Here is an example of a config that will re-encode the stream to H264 without hardware acceleration: ```yaml go2rtc: diff --git a/web/src/components/MsePlayer.js b/web/src/components/MsePlayer.js index a95e8de34..ac0062391 100644 --- a/web/src/components/MsePlayer.js +++ b/web/src/components/MsePlayer.js @@ -217,6 +217,7 @@ class VideoRTC extends HTMLElement { this.video.controls = true; this.video.playsInline = true; this.video.preload = 'auto'; + this.video.muted = true; this.video.style.display = 'block'; // fix bottom margin 4px this.video.style.width = '100%';