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Outgoing call #19

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1 change: 1 addition & 0 deletions .gitignore
Original file line number Diff line number Diff line change
Expand Up @@ -8,6 +8,7 @@ tmp/
htmlcov

.projectile
.env
.venv/
venv/
.mypy_cache/
Expand Down
16 changes: 16 additions & 0 deletions README.md
Original file line number Diff line number Diff line change
@@ -1,3 +1,19 @@
# VoIP Utils

Voice over IP utilities for the [voip integration](https://www.home-assistant.io/integrations/voip/).

## Test outgoing call
Install dependencies from requirements_dev.txt

Set environment variables for source and destination endpoints in .env file
CALL_SRC_USER = "homeassistant"
CALL_SRC_IP = "192.168.1.1"
CALL_SRC_PORT = 5060
CALL_VIA_IP = "192.168.1.1"
CALL_DEST_IP = "192.168.1.2"
CALL_DEST_PORT = 5060
CALL_DEST_USER = "phone"

Run script
python call_example.py

174 changes: 174 additions & 0 deletions call_example.py
Original file line number Diff line number Diff line change
@@ -0,0 +1,174 @@
import asyncio
import logging
import os
import socket
from functools import partial
from pathlib import Path
from typing import Any, Callable, Optional, Set

from dotenv import load_dotenv

from voip_utils.call_phone import VoipCallDatagramProtocol
from voip_utils.sip import (
CallInfo,
CallPhoneDatagramProtocol,
SdpInfo,
SipEndpoint,
get_sip_endpoint,
)
from voip_utils.voip import RtcpDatagramProtocol, RtcpState, RtpDatagramProtocol

_LOGGER = logging.getLogger(__name__)

load_dotenv()


def get_env_int(env_var: str, default_val: int) -> int:
value = os.getenv(env_var)
if value is None:
return default_val
try:
return int(value)
except ValueError:
return default_val


CALL_SRC_USER = os.getenv("CALL_SRC_USER")
CALL_SRC_IP = os.getenv("CALL_SRC_IP", "127.0.0.1")
CALL_SRC_PORT = get_env_int("CALL_SRC_PORT", 5060)
CALL_VIA_IP = os.getenv("CALL_VIA_IP")
CALL_DEST_IP = os.getenv("CALL_DEST_IP", "127.0.0.1")
CALL_DEST_PORT = get_env_int("CALL_DEST_PORT", 5060)
CALL_DEST_USER = os.getenv("CALL_DEST_USER")


RATE = 16000
WIDTH = 2
CHANNELS = 1
RTP_AUDIO_SETTINGS = {
"rate": RATE,
"width": WIDTH,
"channels": CHANNELS,
"sleep_ratio": 0.99,
}


class PreRecordMessageProtocol(RtpDatagramProtocol):
"""Plays a pre-recorded message on a loop."""

def __init__(
self,
file_name: str,
opus_payload_type: int,
message_delay: float = 1.0,
loop_delay: float = 2.0,
rtcp_state: RtcpState | None = None,
) -> None:
"""Set up RTP server."""
super().__init__(
rate=RATE,
width=WIDTH,
channels=CHANNELS,
opus_payload_type=opus_payload_type,
rtcp_state=rtcp_state,
)
self.loop = asyncio.get_running_loop()
self.file_name = file_name
self.message_delay = message_delay
self.loop_delay = loop_delay
self._audio_task: asyncio.Task | None = None
file_path = Path(__file__).parent / self.file_name
self._audio_bytes: bytes = file_path.read_bytes()
_LOGGER.debug("Created PreRecordMessageProtocol")

def on_chunk(self, audio_bytes: bytes) -> None:
"""Handle raw audio chunk."""
_LOGGER.debug("on_chunk")
if self.transport is None:
return

if self._audio_task is None:
self._audio_task = self.loop.create_task(
self._play_message(),
name="voip_not_connected",
)

async def _play_message(self) -> None:
_LOGGER.debug("_play_message")
self.send_audio(
self._audio_bytes,
self.rate,
self.width,
self.channels,
self.addr,
silence_before=self.message_delay,
)

await asyncio.sleep(self.loop_delay)

# Allow message to play again - Only play once for testing
# self._audio_task = None


async def main() -> None:
logging.basicConfig(level=logging.DEBUG)

loop = asyncio.get_event_loop()
source = get_sip_endpoint(
host=CALL_SRC_IP, port=CALL_SRC_PORT, username=CALL_SRC_USER, description=None
)
destination = get_sip_endpoint(
host=CALL_DEST_IP,
port=CALL_DEST_PORT,
username=CALL_DEST_USER,
description=None,
)

# Find free RTP/RTCP ports
rtp_port = 0

while True:
sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
sock.setblocking(False)

# Bind to a random UDP port
sock.bind(("", 0))
_, rtp_port = sock.getsockname()

# Close socket to free port for re-use
sock.close()

# Check that the next port up is available for RTCP
sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
try:
sock.bind(("", rtp_port + 1))

# Will be opened again below
sock.close()

# Found our ports
break
except OSError:
# RTCP port is taken
pass

_, protocol = await loop.create_datagram_endpoint(
lambda: VoipCallDatagramProtocol(
None,
source,
destination,
rtp_port,
lambda call_info, rtcp_state: PreRecordMessageProtocol(
"problem.pcm",
call_info.opus_payload_type,
rtcp_state=rtcp_state,
),
),
local_addr=(CALL_SRC_IP, CALL_SRC_PORT),
)

await protocol.wait_closed()


if __name__ == "__main__":
asyncio.run(main())
Binary file added problem.pcm
Binary file not shown.
1 change: 1 addition & 0 deletions requirements_dev.txt
Original file line number Diff line number Diff line change
Expand Up @@ -6,3 +6,4 @@ isort==5.12.0
mypy==1.1.1
pylint==3.2.5
pytest==7.2.2
python-dotenv==1.0.1
81 changes: 60 additions & 21 deletions tests/test_sip.py
Original file line number Diff line number Diff line change
@@ -1,35 +1,74 @@
"""Test voip_utils SIP functionality."""

from voip_utils.sip import SipDatagramProtocol
from voip_utils.sip import SipEndpoint, get_sip_endpoint


def test_parse_header_for_uri():
endpoint, name = SipDatagramProtocol._parse_uri_header(
None, '"Test Name" <sip:[email protected]>'
)
assert name == "Test Name"
assert endpoint == "sip:[email protected]"
endpoint = SipEndpoint('"Test Name" <sip:[email protected]>')
assert endpoint.description == "Test Name"
assert endpoint.uri == "sip:[email protected]"
assert endpoint.username == "12345"
assert endpoint.host == "example.com"
assert endpoint.port == 5060


def test_parse_header_for_uri_no_name():
endpoint, name = SipDatagramProtocol._parse_uri_header(
None, "sip:[email protected]"
)
assert name is None
assert endpoint == "sip:[email protected]"
endpoint = SipEndpoint("sip:[email protected]")
assert endpoint.description is None
assert endpoint.uri == "sip:[email protected]"


def test_parse_header_for_uri_sips():
endpoint, name = SipDatagramProtocol._parse_uri_header(
None, '"Test Name" <sips:[email protected]>'
)
assert name == "Test Name"
assert endpoint == "sips:[email protected]"
endpoint = SipEndpoint('"Test Name" <sips:[email protected]>')
assert endpoint.description == "Test Name"
assert endpoint.uri == "sips:[email protected]"


def test_parse_header_for_uri_no_space_name():
endpoint, name = SipDatagramProtocol._parse_uri_header(
None, "Test <sip:[email protected]>"
)
assert name == "Test"
assert endpoint == "sip:[email protected]"
endpoint = SipEndpoint("Test <sip:[email protected]>")
assert endpoint.description == "Test"
assert endpoint.uri == "sip:[email protected]"


def test_parse_header_for_uri_no_username():
endpoint = SipEndpoint("Test <sip:example.com>")
assert endpoint.description == "Test"
assert endpoint.username is None
assert endpoint.uri == "sip:example.com"


def test_get_sip_endpoint():
endpoint = get_sip_endpoint("example.com")
assert endpoint.host == "example.com"
assert endpoint.port == 5060
assert endpoint.description is None
assert endpoint.username is None
assert endpoint.uri == "sip:example.com"


def test_get_sip_endpoint_with_username():
endpoint = get_sip_endpoint("example.com", username="test")
assert endpoint.host == "example.com"
assert endpoint.port == 5060
assert endpoint.description is None
assert endpoint.username == "test"
assert endpoint.uri == "sip:[email protected]"


def test_get_sip_endpoint_with_description():
endpoint = get_sip_endpoint("example.com", description="Test Endpoint")
assert endpoint.host == "example.com"
assert endpoint.port == 5060
assert endpoint.description == "Test Endpoint"
assert endpoint.username is None
assert endpoint.uri == "sip:example.com"
assert endpoint.sip_header == '"Test Endpoint" <sip:example.com>'


def test_get_sip_endpoint_with_scheme():
endpoint = get_sip_endpoint("example.com", scheme="sips")
assert endpoint.host == "example.com"
assert endpoint.port == 5060
assert endpoint.description is None
assert endpoint.username is None
assert endpoint.uri == "sips:example.com"
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