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call_example.py
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call_example.py
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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())