Append a 1 to all go2rtc ports to avoid port conflicts (#129881)

homeassistant/components/go2rtc/__init__.py

@@ -38,7 +38,7 @@ from homeassistant.helpers.typing import ConfigType
 from homeassistant.util.hass_dict import HassKey
 from homeassistant.util.package import is_docker_env
 
-from .const import CONF_DEBUG_UI, DEBUG_UI_URL_MESSAGE, DEFAULT_URL, DOMAIN
+from .const import CONF_DEBUG_UI, DEBUG_UI_URL_MESSAGE, DOMAIN, HA_MANAGED_URL
 from .server import Server
 
 _LOGGER = logging.getLogger(__name__)
@@ -125,7 +125,7 @@ async def async_setup(hass: HomeAssistant, config: ConfigType) -> bool:
 
         hass.bus.async_listen(EVENT_HOMEASSISTANT_STOP, on_stop)
 
-        url = DEFAULT_URL
+        url = HA_MANAGED_URL
 
     hass.data[_DATA_GO2RTC] = url
     discovery_flow.async_create_flow(

homeassistant/components/go2rtc/const.py

@@ -4,4 +4,5 @@ DOMAIN = "go2rtc"
 
 CONF_DEBUG_UI = "debug_ui"
 DEBUG_UI_URL_MESSAGE = "Url and debug_ui cannot be set at the same time."
-DEFAULT_URL = "http://localhost:1984/"
+HA_MANAGED_API_PORT = 11984
+HA_MANAGED_URL = f"http://localhost:{HA_MANAGED_API_PORT}/"

homeassistant/components/go2rtc/server.py

@@ -11,7 +11,7 @@ from homeassistant.core import HomeAssistant
 from homeassistant.exceptions import HomeAssistantError
 from homeassistant.helpers.aiohttp_client import async_get_clientsession
 
-from .const import DEFAULT_URL
+from .const import HA_MANAGED_API_PORT, HA_MANAGED_URL
 
 _LOGGER = logging.getLogger(__name__)
 _TERMINATE_TIMEOUT = 5
@@ -26,13 +26,14 @@ _RESPAWN_COOLDOWN = 1
 # - Clear default ice servers
 _GO2RTC_CONFIG_FORMAT = r"""
 api:
-  listen: "{api_ip}:1984"
+  listen: "{api_ip}:{api_port}"
 
 rtsp:
   # ffmpeg needs rtsp for opus audio transcoding
-  listen: "127.0.0.1:8554"
+  listen: "127.0.0.1:18554"
 
 webrtc:
+  listen: ":18555/tcp"
   ice_servers: []
 """
 
@@ -52,7 +53,11 @@ def _create_temp_file(api_ip: str) -> str:
     # Set delete=False to prevent the file from being deleted when the file is closed
     # Linux is clearing tmp folder on reboot, so no need to delete it manually
     with NamedTemporaryFile(prefix="go2rtc_", suffix=".yaml", delete=False) as file:
-        file.write(_GO2RTC_CONFIG_FORMAT.format(api_ip=api_ip).encode())
+        file.write(
+            _GO2RTC_CONFIG_FORMAT.format(
+                api_ip=api_ip, api_port=HA_MANAGED_API_PORT
+            ).encode()
+        )
         return file.name
 
 
@@ -113,7 +118,7 @@ class Server:
             raise Go2RTCServerStartError from err
 
         # Check the server version
-        client = Go2RtcRestClient(async_get_clientsession(self._hass), DEFAULT_URL)
+        client = Go2RtcRestClient(async_get_clientsession(self._hass), HA_MANAGED_URL)
         await client.validate_server_version()
 
     async def _log_output(self, process: asyncio.subprocess.Process) -> None:
@@ -173,7 +178,7 @@ class Server:
 
     async def _monitor_api(self) -> None:
         """Raise if the go2rtc process terminates."""
-        client = Go2RtcRestClient(async_get_clientsession(self._hass), DEFAULT_URL)
+        client = Go2RtcRestClient(async_get_clientsession(self._hass), HA_MANAGED_URL)
 
         _LOGGER.debug("Monitoring go2rtc API")
         try:

tests/components/go2rtc/test_server.py

@@ -71,13 +71,14 @@ async def test_server_run_success(
     mock_tempfile.write.assert_called_once_with(
         f"""
 api:
-  listen: "{api_ip}:1984"
+  listen: "{api_ip}:11984"
 
 rtsp:
   # ffmpeg needs rtsp for opus audio transcoding
-  listen: "127.0.0.1:8554"
+  listen: "127.0.0.1:18554"
 
 webrtc:
+  listen: ":18555/tcp"
   ice_servers: []
 """.encode()
     )