…ut, SDP mono, contact IP, and package hygiene (#9)

Addresses all automated review comments from the initial transcription
implementation.

## CLI (`voip/__main__.py`)
- **Entry point**: `main = voip` (was `sip`), `__main__` calls `voip()`
— exposes the top-level group and `-v/--verbose`
- **`ConsoleMessageProcessor`**: Fixed both `request_received` and
`response_received` to forward correct args to the correct `super()`
method (was calling `super().request_received()` in both, dropping args)
- **ImportError message**: Fixed grammar and package name (`libsip` →
`voip[cli]`)

## Transcription (`voip/whisper.py`)
- **Concurrency guard**: Added `_transcribe_task: asyncio.Task | None` —
`audio_received` only spawns a task when none is running; the task
drains all full chunks in a `while` loop before clearing the guard
- **ffmpeg timeout**: Replaced `.run()` with `.run_async()` +
`proc.communicate(timeout=decode_timeout_secs)` — `decode_timeout_secs`
is now enforced; includes proper subprocess cleanup on timeout and
captures stderr for error messages

## SDP / RTP (`voip/call.py`)
- **Mono codec**: `opus/48000/2` → `opus/48000/1` in SDP — consistent
with the Ogg container (1 channel) and ffmpeg `ac=1` decode path
- **Contact IP**: Added `contact_ip` kwarg to `IncomingCall`;
`IncomingCallProtocol` exposes a `_contact_ip` property (default
`None`), overridden in `RegisterProtocol` to return the STUN-discovered
public IP — avoids advertising `0.0.0.0` in SDP

## Tests
- **`test_integration.py`**: Replaced `RegisterProtocol` with a minimal
`_STUNOnlyProtocol` to avoid sending unintended SIP REGISTER to
`stun.l.google.com`
- **`test_main.py`**: Fixed stub key `"sip.whisper"` → `"voip.whisper"`
- **`test_whisper.py`**: Added `pytest.importorskip("whisper")` guard;
updated ffmpeg mocks to `run_async`/`communicate` API

## Packaging / Docs
- **`pyproject.toml`**: Added `ffmpeg-python` to `cli` extra (required
by the `transcribe` command)
- **`README.md`**: Fixed install snippet `libsip[...]` → `voip[...]`
- **`voip/__init__.py`**: Re-exported `SIP`, `Request`, `Response` so
`import voip; voip.SIP` works as shown in the README example

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Co-authored-by: copilot-swe-agent[bot] <[email protected]>
Co-authored-by: codingjoe <[email protected]>
Co-authored-by: Johannes Maron <[email protected]>
Co-authored-by: pre-commit-ci[bot] <66853113+pre-commit-ci[bot]@users.noreply.github.com>

…FC 8760) (#21)

MD5 is cryptographically broken and its use in SIP Digest Auth is
deprecated by RFC 8760, which mandates SHA-256 and SHA-512-256 support.

## Changes

- **`voip/types.py`**: New `DigestAlgorithm` enum covering all RFC 8760
values: `MD5`, `MD5-sess`, `SHA-256`, `SHA-256-sess`, `SHA-512-256`,
`SHA-512-256-sess`
- **`voip/sip/protocol.py`**:
- `digest_response()` now accepts an `algorithm` parameter (default:
`SHA-256`) and dispatches via `hashlib.new()`. Raises `ValueError` for
unrecognised algorithms.
- `response_received()` reads `algorithm` from the server's
`WWW-Authenticate`/`Proxy-Authenticate` challenge and echoes it in the
`Authorization` header — server-advertised algorithm is honoured
end-to-end.
- **`tests/sip/test_protocol.py`**: Updated existing `algorithm="MD5"`
assertion to `algorithm="SHA-256"`; added `TestDigestResponse` with
coverage for correctness, backward-compat MD5, SHA-512-256, algorithm
distinctness, and server algorithm negotiation.

```python
# Server advertises SHA-256 → client responds with SHA-256
challenge = 'Digest realm="example.com", nonce="abc", algorithm="SHA-256"'
# Authorization header echoes: algorithm="SHA-256", response=<sha256-digest>

# No algorithm in challenge → defaults to SHA-256 (RFC 8760)
digest = SessionInitiationProtocol.digest_response(
    username="alice", ******,
    realm="example.com", nonce="abc",
    method="REGISTER", uri="sip:example.com",
    # algorithm defaults to DigestAlgorithm.SHA_256
)
```

MD5 remains supported for servers that explicitly request it.

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)

RFC 3261 §8.1.2 separates the **outbound proxy** (the server a UA
physically connects to) from the **registrar** (§10, whose URI is
derived from the AOR domain). These can differ — e.g.
`proxy.carrier.com` as the TLS endpoint while `carrier.com` is the
registrar domain.

## Changes

### Protocol (`voip/sip/protocol.py`)

- **`SessionInitiationProtocol`**: renamed `server_address` →
`outbound_proxy` (`tuple[str, int] | None`, default `None`); optional
because the UA may connect directly to the registrar with no explicit
proxy configured
- **Class docstring**: added RFC 3261 topology overview explaining the
outbound proxy / registrar / AOR distinction and when they differ
- **`register()`**: docstring and debug logging now explicitly
distinguish proxy from registrar (`"Sending REGISTER via outbound proxy
… to registrar …"`)

```python
# Proxy and registrar on the same host (most common)
SIP(aor="sips:[email protected]", username="alice", ******)

# Proxy differs from the registrar domain
SIP(
    aor="sips:[email protected]",        # registrar → sips:carrier.com
    outbound_proxy=("proxy.carrier.com", 5061),
    username="alice",
    ******,
)
```

### CLI (`voip/__main__.py`)

The `transcribe` command now takes the AOR as a **positional argument**.
All connection parameters are derived from it by default; individual
pieces can be overridden:

```console
voip sip transcribe sips:[email protected] --password secret
voip sip transcribe sip:[email protected]:5060 --password secret
voip sip transcribe sips:[email protected] --proxy proxy.carrier.com --password secret
```

- **`_parse_aor()`**: new helper that extracts `(scheme, user, host,
port)` from a SIP URI
- **Transport auto-detection from port**: port 5060 → plain TCP; any
other port → TLS. `--no-tls` still overrides explicitly.
- **`sip:` scheme** defaults to port 5060 (plain TCP); **`sips:`**
defaults to port 5061 (TLS).
- **`--username`**: optional override for the user part from the AOR
- **`--proxy HOST[:PORT]`**: optional override for the outbound proxy
address when it differs from the registrar domain in the AOR
- **`SIP_TCP_PORT = 5060` / `SIP_TLS_PORT = 5061`**: named constants
replacing magic numbers

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…otocol hardening (#24)

Addresses all unresolved review comments from the call transcription PR
series.

## SRTP (`voip/srtp.py`)
- **Separate send/receive ROC counters** — `_send_roc`/`_last_send_seq`
for encryption, `_recv_roc`/`_last_recv_seq` for decryption, per RFC
3711 §3.3.1
- **Full index estimation on receive** — `_estimate_recv_index()`
implements the RFC 3711 §3.3.1 algorithm (plain arithmetic comparisons,
not modular)
- **Constant-time auth tag comparison** — replaces `!=` with
`hmac.compare_digest` (stdlib) to prevent timing side-channels

## SIP Protocol (`voip/sip/protocol.py`)
- **Remote RTP address selection** — prefers media-level `c=` →
session-level `c=` → SIP peer IP (per RFC 4566 §5.7); port 0 (inactive
media, RFC 4566 §5.14) skips registration and NAT hole-punching
- **Early INVITE guard** — `_initialize_task` stored on
`connection_made`; `_answer()` awaits it only when `_rtp_protocol` is
still `None`, preventing attribute errors without breaking tests
- **Remove `error_received`** — `asyncio.Protocol` has no such callback
(it belongs to `DatagramProtocol`); removed method and its tests

## Audio (`voip/audio.py`)
- **`WhisperModel` injection** — `WhisperCall.model` now accepts a
pre-loaded `WhisperModel` instance in addition to a name string,
enabling sharing across concurrent calls:

```python
shared_model = WhisperModel("base")

class MyCall(WhisperCall):
    model = shared_model  # loaded once, reused per call
```

## Docs / minor
- **STUN docstring** — removed stale reference to `public_address`
(attribute no longer exists on `STUNProtocol`)
- **RTP docstring** — "silently discarded" → "logged at WARNING level
and discarded" to match actual behavior
- **README** — removed inaccurate "Unencrypted SIP or RTP is not
supported" claim

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