[FWeinb]

Warning

Do not merge yet. This waits for an aiortc release on PyPI that accepts av>=14,<20. Home Assistant pins av==19.0.0, and the current aiortc release (1.15.0) requires av<18, so it cannot be installed alongside Home Assistant. aiortc main already has the relaxed pin, and this branch is built and tested against a commit of it. Until the release is out, the manifest requirement aiortc>=1.16.0 cannot be installed anywhere.

Before merging:

• aiortc released with av>=14,<20; check that aiortc>=1.16.0 in manifest.json names that release
• requirements.txt: replace the aiortc commit pin with the release
• requirements.txt: homeassistant==2026.11.0b0 → 2026.11.0
• requirements_test.txt: pytest-homeassistant from PyPI instead of the core 2026.11.0b0 tag archive
• CI green

Summary

Talk to the person at the door from the camera's live view, using Home Assistant 2026.11's two-way audio preview (Labs). It's opt-in with the new Two-way audio option; without it, the camera stays MJPEG as before.

4ff2443  Write out the config flow strings that used common references   (existing bug)
efe7a00  Test with Home Assistant's pytest-home-assistant on 2026.11
8fbb832  Add two-way audio to the door station camera
7c822e5  Add a header and a HACS badge to the README, list actions in a table

With SIP Auto Answer off and Two-way audio on, Home Assistant rings along with the indoor stations and answers when talk is pressed:

sequenceDiagram
    participant Door as Door station / SG-150
    participant HA as Home Assistant
    participant Browser as Live view
    Door->>HA: INVITE
    HA-->>Door: 180 Ringing (ring along)
    Browser->>HA: WebRTC offer with microphone (talk pressed)
    HA-->>Door: 200 OK, then ACK
    Door-)HA: RTP (G.711)
    HA-)Browser: door audio + VP8 video
    Browser-)HA: microphone
    HA-)Door: RTP (G.711)
    Browser->>HA: close live view (last talker)
    HA->>Door: BYE

Loading

custom_components/sg_150/
├── sip/rtp.py      # new: RTP session per call, G.711, jitter buffer, clip + live mixing
├── sip/minisip.py  # sendrecv both ways, PCMU, CANCEL, resend 200 OK until ACK, ring then answer later
├── line.py         # ring along, answer(), ring timeout
├── talkback.py     # new: aiortc WebRTC sessions, one shared VP8 encoder per camera
├── camera.py       # native WebRTC + two_way_audio capability when the option is on
├── button.py       # Hang Up next to each camera; errors when there is no call
└── media_source.py # image view registered in async_setup (2026.11 loads platforms lazily)

Also in this PR:

• Tests move from pytest-homeassistant-custom-component, which doesn't load on 2026.11, to Home Assistant's own pytest-home-assistant (Clarify pytest-home-assistant docstrings and README home-assistant/core#187272).
• The config flow's [%key:common::…%] references are written out. Custom integrations ship translations/*.json as they are, so users saw the raw reference text.
• New options: Two-way audio, and First/Last RTP Port (default 5004–5013).

Evidence

• Before: on HA 2026.11.0b0, the test suite doesn't start (RuntimeError: httpx was already imported). Stored-image URLs return 404 after a restart, until the media browser has been opened.
  After: scripts/test gives 138 passed, 84% coverage. This includes an in-process WebRTC test with a second aiortc peer as the browser:

  ringing call + browser offer without microphone  → video arrives, call keeps ringing
  ringing call + browser offer with microphone     → 200 OK, mic tone reaches the gateway's RTP,
                                                     door audio reaches the browser
  close the last talking session                   → BYE
  

• Hardware: tested by hand on a real SG-150 with a local HA 2026.11.0b0 instance. Ring-along, talking both ways, hanging up, and play_audio all work. The live view shows video while the call is still ringing.

Merge Danger

Door: two-way

Two-way audio is opt-in and off by default. Reverting the PR restores the previous behaviour, with no config migration. Saved options gain three keys, which older versions ignore.

Blast Radius: calls

• Every call changes, not only two-way audio calls. Call audio now goes through a per-call RTP port (5004-5013 by default), not the SIP port. The SG-150 must be able to reach those ports, so firewalls may need updating. Outgoing calls offer sendrecv instead of sendonly.
• Minimum Home Assistant becomes 2026.11.0 (hacs.json). HACS won't offer this version to older installs. Installed by hand on older HA, the camera platform fails to load, because CameraEntityCapabilities doesn't exist there.
• The camera live view becomes WebRTC once the option is on. Encoding costs about a third of a CPU core per camera while someone watches (VP8, 768×576, 15 fps, measured on the HA host).
• The two-way audio API is a Labs preview and may change in later Home Assistant releases.