VoIP 0.1.0__tar.gz

voip-0.1.0/LICENSE

@@ -0,0 +1,24 @@
+BSD 2-Clause License
+
+Copyright (c) 2026, Johannes Maron
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are met:
+
+1. Redistributions of source code must retain the above copyright notice, this
+   list of conditions and the following disclaimer.
+
+2. Redistributions in binary form must reproduce the above copyright notice,
+   this list of conditions and the following disclaimer in the documentation
+   and/or other materials provided with the distribution.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
+FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

voip-0.1.0/PKG-INFO

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+Metadata-Version: 2.4
+Name: VoIP
+Version: 0.1.0
+Summary: Python asyncio library for VoIP calls.
+Keywords: VoIP,SIP,RTP,WebRTC,STUN,asyncio
+Author-email: Johannes Maron <johannes@maron.family>
+Requires-Python: >=3.13
+Description-Content-Type: text/markdown
+Classifier: Development Status :: 3 - Alpha
+Classifier: Environment :: Console
+Classifier: Intended Audience :: Developers
+Classifier: Intended Audience :: Information Technology
+Classifier: Intended Audience :: Telecommunications Industry
+Classifier: License :: OSI Approved :: BSD License
+Classifier: Operating System :: OS Independent
+Classifier: Programming Language :: Python
+Classifier: Programming Language :: Python :: 3
+Classifier: Programming Language :: Python :: 3 :: Only
+Classifier: Programming Language :: Python :: 3.13
+Classifier: Programming Language :: Python :: 3.14
+Classifier: Framework :: AsyncIO
+Classifier: Topic :: Communications :: Internet Phone
+Classifier: Topic :: Communications :: Telephony
+Classifier: Topic :: Multimedia :: Sound/Audio
+Classifier: Topic :: Multimedia :: Sound/Audio :: Speech
+Classifier: Topic :: Scientific/Engineering :: Artificial Intelligence
+Classifier: Topic :: Software Development :: Libraries
+Classifier: Topic :: System :: Networking
+Classifier: Topic :: Home Automation
+License-File: LICENSE
+Requires-Dist: cryptography
+Requires-Dist: faster-whisper ; extra == "audio"
+Requires-Dist: numpy ; extra == "audio"
+Requires-Dist: av ; extra == "audio"
+Requires-Dist: click ; extra == "cli"
+Requires-Dist: pygments ; extra == "cli"
+Requires-Dist: faster-whisper ; extra == "cli"
+Requires-Dist: numpy ; extra == "cli"
+Requires-Dist: av ; extra == "cli"
+Requires-Dist: Pygments ; extra == "pygments"
+Project-URL: Changelog, https://github.com/codingjoe/VoIP/releases
+Project-URL: Project-URL, https://github.com/codingjoe/VoIP
+Provides-Extra: audio
+Provides-Extra: cli
+Provides-Extra: pygments
+
+<p align="center">
+  <picture>
+    <source media="(prefers-color-scheme: dark)" srcset="https://github.com/codingjoe/VoIP/raw/main/docs/images/logo-dark.svg">
+    <source media="(prefers-color-scheme: light)" srcset="https://github.com/codingjoe/VoIP/raw/main/docs/images/logo-light.svg">
+    <img alt="Python VoIP" src="https://github.com/codingjoe/VoIP/raw/main/docs/images/logo-light.svg">
+  </picture>
+<br>
+  <a href="https://github.com/codingjoe">Documentation</a> |
+  <a href="https://github.com/codingjoe/VoIP/issues/new/choose">Issues</a> |
+  <a href="https://github.com/codingjoe/VoIP/releases">Changelog</a> |
+  <a href="https://github.com/sponsors/codingjoe">Funding</a> 💚
+</p>
+
+# Python VoIP library
+
+> [!WARNING]
+> This library is in early development and may contain breaking changes. Use with caution.
+
+Python asyncio library for SIP telephony ([RFC 3261](https://tools.ietf.org/html/rfc3261)).
+
+All signalling uses **SIP over TLS** (SIPS, RFC 3261 §26) and all media is
+protected with **SRTP** ([RFC 3711](https://tools.ietf.org/html/rfc3711))
+using the `AES_CM_128_HMAC_SHA1_80` cipher suite with SDES key exchange
+([RFC 4568](https://tools.ietf.org/html/rfc4568)).
+
+## Setup
+
+```console
+pip install voip[audio,cli,pygments]
+```
+
+## Usage
+
+### CLI
+
+Answer calls and transcribe them live from the terminal:
+
+```console
+voip sip transcribe sips:alice@sip.example.com --password secret
+```
+
+### Python API
+
+Subclass `WhisperCall` and override `transcription_received` to handle results.
+Pass it as `call_class` when answering an incoming call:
+
+```python
+import asyncio
+import ssl
+from voip.audio import WhisperCall
+from voip.sip.protocol import SIP
+
+
+class MyCall(WhisperCall):
+    def transcription_received(self, text: str) -> None:
+        print(f"[{self.caller}] {text}")
+
+
+class MySession(SIP):
+    def call_received(self, request) -> None:
+        asyncio.create_task(self.answer(request=request, call_class=MyCall))
+
+
+async def main():
+    loop = asyncio.get_running_loop()
+    ssl_context = ssl.create_default_context()
+    await loop.create_connection(
+        lambda: MySession(
+            aor="sips:alice@example.com",
+            username="alice",
+            password="secret",
+        ),
+        host="sip.example.com",
+        port=5061,
+        ssl=ssl_context,
+    )
+    await asyncio.Future()
+
+
+asyncio.run(main())
+```
+
+For raw audio access without transcription, subclass `AudioCall` and override
+`audio_received(self, audio: np.ndarray)` instead.
+

voip-0.1.0/README.md

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+<p align="center">
+  <picture>
+    <source media="(prefers-color-scheme: dark)" srcset="https://github.com/codingjoe/VoIP/raw/main/docs/images/logo-dark.svg">
+    <source media="(prefers-color-scheme: light)" srcset="https://github.com/codingjoe/VoIP/raw/main/docs/images/logo-light.svg">
+    <img alt="Python VoIP" src="https://github.com/codingjoe/VoIP/raw/main/docs/images/logo-light.svg">
+  </picture>
+<br>
+  <a href="https://github.com/codingjoe">Documentation</a> |
+  <a href="https://github.com/codingjoe/VoIP/issues/new/choose">Issues</a> |
+  <a href="https://github.com/codingjoe/VoIP/releases">Changelog</a> |
+  <a href="https://github.com/sponsors/codingjoe">Funding</a> 💚
+</p>
+
+# Python VoIP library
+
+> [!WARNING]
+> This library is in early development and may contain breaking changes. Use with caution.
+
+Python asyncio library for SIP telephony ([RFC 3261](https://tools.ietf.org/html/rfc3261)).
+
+All signalling uses **SIP over TLS** (SIPS, RFC 3261 §26) and all media is
+protected with **SRTP** ([RFC 3711](https://tools.ietf.org/html/rfc3711))
+using the `AES_CM_128_HMAC_SHA1_80` cipher suite with SDES key exchange
+([RFC 4568](https://tools.ietf.org/html/rfc4568)).
+
+## Setup
+
+```console
+pip install voip[audio,cli,pygments]
+```
+
+## Usage
+
+### CLI
+
+Answer calls and transcribe them live from the terminal:
+
+```console
+voip sip transcribe sips:alice@sip.example.com --password secret
+```
+
+### Python API
+
+Subclass `WhisperCall` and override `transcription_received` to handle results.
+Pass it as `call_class` when answering an incoming call:
+
+```python
+import asyncio
+import ssl
+from voip.audio import WhisperCall
+from voip.sip.protocol import SIP
+
+
+class MyCall(WhisperCall):
+    def transcription_received(self, text: str) -> None:
+        print(f"[{self.caller}] {text}")
+
+
+class MySession(SIP):
+    def call_received(self, request) -> None:
+        asyncio.create_task(self.answer(request=request, call_class=MyCall))
+
+
+async def main():
+    loop = asyncio.get_running_loop()
+    ssl_context = ssl.create_default_context()
+    await loop.create_connection(
+        lambda: MySession(
+            aor="sips:alice@example.com",
+            username="alice",
+            password="secret",
+        ),
+        host="sip.example.com",
+        port=5061,
+        ssl=ssl_context,
+    )
+    await asyncio.Future()
+
+
+asyncio.run(main())
+```
+
+For raw audio access without transcription, subclass `AudioCall` and override
+`audio_received(self, audio: np.ndarray)` instead.

voip-0.1.0/pyproject.toml

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+[build-system]
+requires = ["flit_core>=3.2", "flit_scm", "wheel"]
+build-backend = "flit_scm:buildapi"
+
+[project]
+name = "VoIP"
+authors = [
+  { name = "Johannes Maron", email = "johannes@maron.family" },
+]
+readme = "README.md"
+license = { file = "LICENSE" }
+keywords = ["VoIP", "SIP", "RTP", "WebRTC", "STUN", "asyncio"]
+dynamic = ["version", "description"]
+classifiers = [
+  # https://pypi.org/classifiers/
+  "Development Status :: 3 - Alpha",
+  "Environment :: Console",
+  "Intended Audience :: Developers",
+  "Intended Audience :: Information Technology",
+  "Intended Audience :: Telecommunications Industry",
+  "License :: OSI Approved :: BSD License",
+  "Operating System :: OS Independent",
+  "Programming Language :: Python",
+  "Programming Language :: Python :: 3",
+  "Programming Language :: Python :: 3 :: Only",
+  "Programming Language :: Python :: 3.13",
+  "Programming Language :: Python :: 3.14",
+  "Framework :: AsyncIO",
+  "Topic :: Communications :: Internet Phone",
+  "Topic :: Communications :: Telephony",
+  "Topic :: Multimedia :: Sound/Audio",
+  "Topic :: Multimedia :: Sound/Audio :: Speech",
+  "Topic :: Scientific/Engineering :: Artificial Intelligence",
+  "Topic :: Software Development :: Libraries",
+  "Topic :: System :: Networking",
+  "Topic :: Home Automation",
+]
+requires-python = ">=3.13"
+dependencies = ["cryptography"]
+
+[project.optional-dependencies]
+audio = ["faster-whisper", "numpy", "av"]
+cli = ["click", "pygments", "faster-whisper", "numpy", "av"]
+pygments = ["Pygments"]
+
+[project.scripts]
+voip = "voip.__main__:main"
+
+[project.entry-points."pygments.lexers"]
+sip = "voip.sip.lexers:SIPLexer"
+sdp = "voip.sdp.lexers:SDPLexer"
+
+[tool.flit.module]
+name = "voip"
+
+[dependency-groups]
+dev = [
+  { include-group = "test" },
+]
+test = [
+  "pytest",
+  "pytest-cov",
+  "pytest-asyncio>=1.3.0",
+]
+
+[project.urls]
+Project-URL = "https://github.com/codingjoe/VoIP"
+Changelog = "https://github.com/codingjoe/VoIP/releases"
+
+[tool.setuptools_scm]
+write_to = "voip/_version.py"
+
+[tool.pytest.ini_options]
+minversion = "6.0"
+addopts = "--cov --strict-markers --cov-report=xml --cov-report=term"
+asyncio_mode = "auto"
+testpaths = ["tests"]
+markers = [
+    "integration: marks tests as integration tests requiring network access (deselect with '-m \"not integration\"')",
+]
+
+[tool.coverage.run]
+source = ["voip"]
+
+[tool.coverage.report]
+show_missing = true
+
+[tool.ruff]
+src = ["voip", "tests"]
+
+[tool.ruff.lint]
+select = [
+    "E", # pycodestyle errors
+    "W", # pycodestyle warnings
+    "F", # pyflakes
+    "I", # isort
+    "S", # flake8-bandit
+    "D", # pydocstyle
+    "UP", # pyupgrade
+    "B", # flake8-bugbear
+    "C", # flake8-comprehensions
+]
+
+ignore = ["B904", "D1", "E501", "S101"]
+
+[tool.ruff.lint.isort]
+combine-as-imports = true
+split-on-trailing-comma = true
+section-order = ["future", "standard-library", "third-party", "first-party", "local-folder"]
+force-wrap-aliases = true
+
+[tool.ruff.lint.pydocstyle]
+convention = "pep257"

voip-0.1.0/voip/__init__.py

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+"""Python asyncio library for VoIP calls."""
+
+from . import _version
+
+__version__ = _version.version
+VERSION = _version.version_tuple

voip-0.1.0/voip/__main__.py

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+#!/usr/bin/env python3
+import asyncio
+import dataclasses
+import logging
+import ssl
+import time
+
+from voip.sip import messages
+
+try:
+    import click
+    from pygments import formatters, highlight
+
+    from voip.sip.lexers import SIPLexer
+except ImportError as e:
+    raise ImportError(
+        "The VoIP CLI requires extra dependencies. Install via `pip install voip[cli]`."
+    ) from e
+
+
+class ConsoleMessageProcessor:
+    """Protocol mixin that prints messages to stdout."""
+
+    def request_received(self, request: messages.Request, addr: tuple[str, int]):
+        self.pprint(request)
+        super().request_received(request, addr)
+
+    def response_received(self, response: messages.Response, addr: tuple[str, int]):
+        self.pprint(response)
+        super().response_received(response, addr)
+
+    def send(self, message) -> None:
+        """Send a message and print it to stdout."""
+        self.pprint(message)
+        super().send(message)
+
+    def pprint(self, msg):
+        """Pretty print the message."""
+        transport = getattr(self, "transport", None)
+        addr = transport.get_extra_info("peername") if transport else None
+        if addr:
+            host = f"[{addr[0]}]" if ":" in addr[0] else addr[0]
+            host = click.style(host, fg="green", bold=True)
+            port = click.style(str(addr[1]), fg="yellow", bold=True)
+            prefix = f"{host}:{port} - - [{time.asctime()}]"
+        else:
+            prefix = f"[unknown] - - [{time.asctime()}]"
+        pretty_msg = highlight(str(msg), SIPLexer(), formatters.TerminalFormatter())
+        click.echo(f"{prefix} {pretty_msg}")
+
+
+@click.group()
+@click.option("-v", "--verbose", count=True, help="Increase verbosity.")
+@click.pass_context
+def voip(ctx, verbose):
+    """VoIP command line interface."""
+    ctx.ensure_object(dict)
+    ctx.obj["verbose"] = verbose
+    logging.basicConfig(
+        level=max(10, 10 * (4 - verbose)),
+        format="%(levelname)s: [%(asctime)s] (%(name)s) %(message)s",
+        handlers=[logging.StreamHandler()],
+    )
+    logging.getLogger("voip").setLevel(max(10, 10 * (3 - verbose)))
+
+
+@voip.group()
+def sip():
+    """Session Initiation Protocol (SIP)."""
+
+
+main = voip
+
+#: Standard SIP/TCP port — plain text, no TLS (RFC 3261 §18.2).
+SIP_TCP_PORT = 5060
+#: Standard SIP/TLS port (RFC 3261 §26.2.2).
+SIP_TLS_PORT = 5061
+
+
+def _parse_aor(value: str) -> tuple[str, str, str, int | None]:
+    """Parse a SIP URI into ``(scheme, user, host, port)``.
+
+    The port is ``None`` when not present in the URI.
+
+    Examples::
+
+        >>> _parse_aor("sip:alice@example.com")
+        ('sip', 'alice', 'example.com', None)
+        >>> _parse_aor("sips:+15551234567@carrier.com:5061")
+        ('sips', '+15551234567', 'carrier.com', 5061)
+    """
+    scheme, _, rest = value.partition(":")
+    if not scheme or not rest:
+        raise click.BadParameter(
+            f"Invalid SIP URI: {value!r}. Expected sip[s]:user@host[:port]."
+        )
+    user_part, _, hostport = rest.partition("@")
+    if not hostport:
+        raise click.BadParameter(f"Invalid SIP URI: {value!r}. Missing user@host part.")
+    host, _, port_str = hostport.partition(":")
+    if not host:
+        raise click.BadParameter(f"Invalid SIP URI: {value!r}. Missing host.")
+    port: int | None = int(port_str) if port_str else None
+    return scheme, user_part, host, port
+
+
+def _parse_hostport(
+    ctx, param, value: str, default_port: int = 5061
+) -> tuple[str, int]:
+    """Parse ``HOST[:PORT]`` into a ``(host, port)`` tuple."""
+    host, _, port_str = value.rpartition(":")
+    if not host:
+        return value, default_port
+    try:
+        return host, int(port_str)
+    except ValueError:
+        raise click.BadParameter(f"Invalid port in {value!r}.", param=param) from None
+
+
+def _parse_stun_server(ctx, param, value: str | None) -> tuple[str, int] | None:
+    """Parse the --stun-server option; return None when the value is 'none'."""
+    if value is None or value.lower() == "none":
+        return None
+    return _parse_hostport(ctx, param, value, default_port=3478)
+
+
+# Keep the old name as an alias so existing internal callers still work.
+_parse_server = _parse_hostport
+
+
+@sip.command()
+@click.argument("aor", metavar="AOR", envvar="SIP_AOR")
+@click.option(
+    "--model",
+    default="base",
+    envvar="WHISPER_MODEL",
+    show_default=True,
+    help="Whisper model size.",
+)
+@click.option(
+    "--password",
+    envvar="SIP_PASSWORD",
+    required=True,
+    help="SIP password (not parsed from AOR for security).",
+)
+@click.option(
+    "--username",
+    envvar="SIP_USERNAME",
+    default=None,
+    help="Override SIP username (defaults to user part of AOR).",
+)
+@click.option(
+    "--proxy",
+    envvar="SIP_PROXY",
+    default=None,
+    metavar="HOST[:PORT]",
+    help=(
+        "Outbound proxy address (RFC 3261 §8.1.2). "
+        "Defaults to the host and port from AOR. "
+        "Use this when the proxy differs from the registrar domain."
+    ),
+)
+@click.option(
+    "--stun-server",
+    envvar="STUN_SERVER",
+    default="stun.cloudflare.com:3478",
+    show_default=True,
+    metavar="HOST[:PORT]",
+    callback=_parse_stun_server,
+    is_eager=False,
+    help="STUN server for RTP NAT traversal (use 'none' to disable).",
+)
+@click.option(
+    "--no-tls",
+    is_flag=True,
+    default=False,
+    help=(
+        "Force plain TCP — skips TLS. "
+        "Auto-selected when port 5060 is used; explicit flag overrides any port."
+    ),
+)
+@click.option(
+    "--no-verify-tls",
+    is_flag=True,
+    default=False,
+    help="Disable TLS certificate verification (insecure; for testing only).",
+)
+@click.pass_context
+def transcribe(
+    ctx, aor, model, password, username, proxy, stun_server, no_tls, no_verify_tls
+):
+    r"""Register with a SIP carrier and transcribe incoming calls via Whisper.
+
+    AOR is a SIP Address of Record URI identifying the account to register,
+    e.g. ``sips:alice@carrier.example.com`` or ``sip:alice@carrier.example.com:5060``.
+
+    \b
+    Transport selection (overridable with --no-tls):
+      sips: URI or port 5061  →  TLS (default)
+      sip:  URI or port 5060  →  plain TCP
+
+    \b
+    Examples:
+      voip sip transcribe sips:alice@sip.example.com --password secret
+      voip sip transcribe sip:alice@sip.example.com:5060 --password secret
+      voip sip transcribe sips:alice@carrier.com --proxy proxy.carrier.com --password secret
+    """
+    from voip.sip.protocol import SIP
+
+    from .audio import WhisperCall  # noqa: PLC0415
+
+    try:
+        scheme, aor_user, aor_host, aor_port = _parse_aor(aor)
+    except click.BadParameter as exc:
+        raise click.BadParameter(str(exc), param_hint="AOR") from exc
+
+    effective_username = username or aor_user
+
+    # Determine outbound proxy address: --proxy overrides, otherwise use AOR host.
+    if proxy is not None:
+        proxy_addr = _parse_hostport(ctx, None, proxy)
+    else:
+        # Default port: SIP_TCP_PORT for sip scheme, SIP_TLS_PORT for sips.
+        default_port = SIP_TCP_PORT if scheme == "sip" else SIP_TLS_PORT
+        port = aor_port if aor_port is not None else default_port
+        proxy_addr = (aor_host, port)
+
+    # Transport: port 5060 (SIP_TCP_PORT) → plain TCP; any other port → TLS.
+    # --no-tls always overrides this auto-detection.
+    use_tls = not no_tls and proxy_addr[1] != SIP_TCP_PORT
+
+    # The AOR stored in the protocol must NOT include the port
+    # (AOR is scheme:user@host per RFC 3261 §10).
+    normalized_aor = f"{scheme}:{effective_username}@{aor_host}"
+
+    verbose = ctx.obj.get("verbose", 0)
+
+    # Capture the CLI model arg as the dataclass field default so that the
+    # class can still be passed as a plain type to SIP.answer().
+    _model = model
+
+    @dataclasses.dataclass
+    class TranscribingCall(WhisperCall):
+        """WhisperCall with the CLI-selected model and console output."""
+
+        model: str = _model
+
+        def transcription_received(self, text: str) -> None:
+            click.echo(click.style(text, fg="green", bold=True))
+
+    # Mix in ConsoleMessageProcessor only at maximum verbosity (-vvv) so that
+    # normal operation is not flooded with protocol-level message dumps.
+    bases = (ConsoleMessageProcessor, SIP) if verbose >= 3 else (SIP,)
+
+    class TranscribeSession(*bases):
+        def call_received(self, request) -> None:
+            self.ringing(request=request)
+            asyncio.create_task(
+                self.answer(request=request, call_class=TranscribingCall)
+            )
+
+    async def run():
+        loop = asyncio.get_running_loop()
+        if use_tls:
+            ssl_context = ssl.create_default_context()
+            if no_verify_tls:
+                ssl_context.check_hostname = False
+                ssl_context.verify_mode = ssl.CERT_NONE
+        else:
+            ssl_context = None
+        await loop.create_connection(
+            lambda: TranscribeSession(
+                outbound_proxy=proxy_addr,
+                aor=normalized_aor,
+                username=effective_username,
+                password=password,
+                rtp_stun_server_address=stun_server,
+            ),
+            host=proxy_addr[0],
+            port=proxy_addr[1],
+            ssl=ssl_context,
+        )
+        await asyncio.Future()
+
+    try:
+        asyncio.run(run())
+    except KeyboardInterrupt:
+        pass
+
+
+if __name__ == "__main__":  # pragma: no cover
+    main()

voip-0.1.0/voip/_version.py

@@ -0,0 +1,34 @@
+# file generated by setuptools-scm
+# don't change, don't track in version control
+
+__all__ = [
+    "__version__",
+    "__version_tuple__",
+    "version",
+    "version_tuple",
+    "__commit_id__",
+    "commit_id",
+]
+
+TYPE_CHECKING = False
+if TYPE_CHECKING:
+    from typing import Tuple
+    from typing import Union
+
+    VERSION_TUPLE = Tuple[Union[int, str], ...]
+    COMMIT_ID = Union[str, None]
+else:
+    VERSION_TUPLE = object
+    COMMIT_ID = object
+
+version: str
+__version__: str
+__version_tuple__: VERSION_TUPLE
+version_tuple: VERSION_TUPLE
+commit_id: COMMIT_ID
+__commit_id__: COMMIT_ID
+
+__version__ = version = '0.1.0'
+__version_tuple__ = version_tuple = (0, 1, 0)
+
+__commit_id__ = commit_id = 'g1a0b5c6d1'