Files
ha-sip-notifier/sip-notifier/sip_service.py

331 lines
10 KiB
Python

#!/usr/bin/env python3
"""SIP Voice Notifier Add-on Service."""
import json
import logging
import os
import tempfile
import time
from urllib.parse import urlparse
import requests
from flask import Flask, request, jsonify
from pydub import AudioSegment
import pjsua2 as pj
# Configure logging
logging.basicConfig(
level=logging.INFO,
format='%(asctime)s - %(name)s - %(levelname)s - %(message)s'
)
_LOGGER = logging.getLogger(__name__)
app = Flask(__name__)
# Global config
CONFIG = {}
DEFAULT_SAMPLE_RATE = 8000
class CallHandler(pj.Call):
"""Handle SIP call events."""
def __init__(self, account, call_id=pj.PJSUA_INVALID_ID):
pj.Call.__init__(self, account, call_id)
self.player = None
self.audio_file = None
self.connected = False
def onCallState(self, prm):
"""Called when call state changes."""
ci = self.getInfo()
_LOGGER.info(f"Call state: {ci.stateText}")
if ci.state == pj.PJSIP_INV_STATE_CONFIRMED:
_LOGGER.info("Call connected! Playing audio...")
self.connected = True
self.play_audio()
elif ci.state == pj.PJSIP_INV_STATE_DISCONNECTED:
_LOGGER.info("Call disconnected")
def onCallMediaState(self, prm):
"""Called when media state changes."""
ci = self.getInfo()
for mi in ci.media:
if mi.type == pj.PJMEDIA_TYPE_AUDIO and mi.status == pj.PJSUA_CALL_MEDIA_ACTIVE:
aud_med = self.getAudioMedia(mi.index)
try:
pj.Endpoint.instance().audDevManager().getPlaybackDevMedia().startTransmit(aud_med)
aud_med.startTransmit(pj.Endpoint.instance().audDevManager().getPlaybackDevMedia())
except Exception as e:
_LOGGER.warning(f"Audio routing warning: {e}")
def play_audio(self):
"""Play the audio file into the call."""
if not self.audio_file:
_LOGGER.error("No audio file specified")
return
try:
self.player = pj.AudioMediaPlayer()
self.player.createPlayer(self.audio_file, pj.PJMEDIA_FILE_NO_LOOP)
ci = self.getInfo()
for mi in ci.media:
if mi.type == pj.PJMEDIA_TYPE_AUDIO and mi.status == pj.PJSUA_CALL_MEDIA_ACTIVE:
aud_med = self.getAudioMedia(mi.index)
self.player.startTransmit(aud_med)
_LOGGER.info(f"Playing: {self.audio_file}")
break
except Exception as e:
_LOGGER.error(f"Error playing audio: {e}", exc_info=True)
def set_audio_file(self, audio_file: str):
"""Set the audio file to play."""
self.audio_file = audio_file
class SIPAccount(pj.Account):
"""SIP Account handler."""
def __init__(self):
pj.Account.__init__(self)
def onIncomingCall(self, prm):
"""Reject incoming calls."""
call = CallHandler(self, prm.callId)
call_param = pj.CallOpParam()
call_param.statusCode = 486
call.answer(call_param)
def download_and_convert_audio(url: str) -> str:
"""Download and convert audio to WAV."""
parsed_url = urlparse(url)
extension = os.path.splitext(parsed_url.path)[1] or ".mp3"
with tempfile.NamedTemporaryFile(delete=False, suffix=extension) as tmp:
download_path = tmp.name
response = requests.get(url, stream=True, timeout=30)
response.raise_for_status()
with open(download_path, 'wb') as f:
for chunk in response.iter_content(chunk_size=8192):
f.write(chunk)
_LOGGER.debug(f"Downloaded: {download_path}")
try:
audio = AudioSegment.from_file(download_path)
audio = audio.set_frame_rate(DEFAULT_SAMPLE_RATE)
audio = audio.set_channels(1)
audio = audio.set_sample_width(2)
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp:
wav_path = tmp.name
audio.export(wav_path, format="wav")
_LOGGER.debug(f"Converted: {wav_path}")
os.remove(download_path)
return wav_path
except Exception as e:
if os.path.exists(download_path):
os.remove(download_path)
raise Exception(f"Audio conversion failed: {e}")
def generate_tts_audio(message: str) -> str:
"""Generate TTS audio."""
try:
from gtts import gTTS
with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp:
mp3_path = tmp.name
tts = gTTS(text=message, lang='en')
tts.save(mp3_path)
audio = AudioSegment.from_mp3(mp3_path)
audio = audio.set_frame_rate(DEFAULT_SAMPLE_RATE)
audio = audio.set_channels(1)
audio = audio.set_sample_width(2)
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp:
wav_path = tmp.name
audio.export(wav_path, format="wav")
os.remove(mp3_path)
return wav_path
except ImportError:
raise Exception("gTTS not available")
def place_sip_call(destination: str, audio_file: str, duration: int):
"""Place a SIP call."""
ep = pj.Endpoint()
ep.libCreate()
ep_cfg = pj.EpConfig()
ep_cfg.logConfig.level = 3
ep_cfg.logConfig.consoleLevel = 0
ep.libInit(ep_cfg)
transport_cfg = pj.TransportConfig()
transport_cfg.port = 0
ep.transportCreate(pj.PJSIP_TRANSPORT_UDP, transport_cfg)
ep.libStart()
_LOGGER.info("PJSIP started")
try:
acc = SIPAccount()
acc_cfg = pj.AccountConfig()
sip_user = CONFIG['sip_user']
sip_server = CONFIG['sip_server']
sip_password = CONFIG['sip_password']
if not sip_user.startswith('sip:'):
sip_uri = f"sip:{sip_user}@{sip_server}"
else:
sip_uri = sip_user
acc_cfg.idUri = sip_uri
acc_cfg.regConfig.registrarUri = f"sip:{sip_server}"
cred = pj.AuthCredInfo("digest", "*", sip_user, 0, sip_password)
acc_cfg.sipConfig.authCreds.append(cred)
acc.create(acc_cfg)
_LOGGER.info(f"Account created: {sip_uri}")
time.sleep(2)
if not destination.startswith('sip:'):
dest_uri = f"sip:{destination}@{sip_server}"
else:
dest_uri = destination
_LOGGER.info(f"Calling: {dest_uri}")
call = CallHandler(acc)
call.set_audio_file(audio_file)
call_param = pj.CallOpParam()
call_param.opt.audioCount = 1
call_param.opt.videoCount = 0
call.makeCall(dest_uri, call_param)
wait_time = 0
while not call.connected and wait_time < 10:
time.sleep(0.5)
wait_time += 0.5
if not call.connected:
_LOGGER.warning("Call did not connect")
_LOGGER.info(f"Call active for {duration}s")
time.sleep(duration)
_LOGGER.info("Hanging up...")
hangup_param = pj.CallOpParam()
call.hangup(hangup_param)
time.sleep(1)
except Exception as e:
_LOGGER.error(f"Call error: {e}", exc_info=True)
raise
finally:
try:
ep.libDestroy()
except Exception as e:
_LOGGER.warning(f"Cleanup warning: {e}")
@app.route('/health', methods=['GET'])
def health():
"""Health check endpoint."""
return jsonify({'status': 'ok'}), 200
@app.route('/send_notification', methods=['POST'])
def handle_send_notification():
"""
Handle send_notification service call.
This endpoint is called by Home Assistant when the service is invoked.
"""
try:
data = request.json
destination = data.get('destination')
audio_url = data.get('audio_url')
message = data.get('message')
duration = data.get('duration', CONFIG.get('default_duration', 30))
if not destination:
return jsonify({'error': 'destination required'}), 400
if not audio_url and not message:
return jsonify({'error': 'audio_url or message required'}), 400
temp_files = []
try:
# Prepare audio
if audio_url:
_LOGGER.info(f"Downloading: {audio_url}")
audio_file = download_and_convert_audio(audio_url)
temp_files.append(audio_file)
elif message:
_LOGGER.info(f"Generating TTS: {message}")
audio_file = generate_tts_audio(message)
temp_files.append(audio_file)
# Place call
_LOGGER.info(f"Calling: {destination}")
place_sip_call(destination, audio_file, duration)
return jsonify({'status': 'success'}), 200
finally:
# Cleanup
for f in temp_files:
try:
if os.path.exists(f):
os.remove(f)
except Exception as e:
_LOGGER.warning(f"Cleanup failed: {e}")
except Exception as e:
_LOGGER.error(f"Request error: {e}", exc_info=True)
return jsonify({'error': str(e)}), 500
if __name__ == '__main__':
# Load config from Home Assistant add-on options
options_file = '/data/options.json'
if os.path.exists(options_file):
with open(options_file, 'r') as f:
CONFIG = json.load(f)
_LOGGER.info("Config loaded from options.json")
else:
_LOGGER.warning("No options.json found, using environment variables")
CONFIG = {
'sip_server': os.getenv('SIP_SERVER', ''),
'sip_user': os.getenv('SIP_USER', ''),
'sip_password': os.getenv('SIP_PASSWORD', ''),
'default_duration': int(os.getenv('DEFAULT_DURATION', 30))
}
_LOGGER.info("SIP Voice Notifier ready - service will be auto-registered by Supervisor")
_LOGGER.info("Starting service on port 8099")
app.run(host='0.0.0.0', port=8099, debug=False)