Aymerick/ha-sip-notifier
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Replace PJSIP with pyVoIP - simpler, pure Python SIP library (v2.0.3)

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OpenClaw Bot
2026-02-08 14:34:46 +01:00
parent 8e54da29fe
commit 76136b5566
4 changed files with 51 additions and 196 deletions
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56
sip-notifier/Dockerfile
View File

@@ -1,62 +1,14 @@
ARG BUILD_FROM
FROM $BUILD_FROM as builder
# Install build dependencies
RUN apk add --no-cache \
git \
build-base \
python3-dev \
linux-headers \
openssl-dev \
alsa-lib-dev \
opus-dev \
speex-dev \
speexdsp-dev \
py3-pip
# Build PJSIP
WORKDIR /tmp
RUN git clone --depth 1 --branch 2.14.1 https://github.com/pjsip/pjproject.git && \
cd pjproject && \
./configure \
--prefix=/opt/pjsip \
--enable-shared \
--disable-video \
--disable-opencore-amr \
--disable-silk \
--disable-opus \
--disable-resample \
--disable-speex-aec \
--disable-g711-codec \
--disable-l16-codec \
--disable-g722-codec && \
make dep && \
make && \
make install && \
cd pjsip-apps/src/python && \
python3 setup.py build && \
python3 setup.py install --prefix=/opt/pjsip
# Final stage
FROM $BUILD_FROM
# Install runtime dependencies only
# Install system dependencies
RUN apk add --no-cache \
python3 \
py3-pip \
ffmpeg \
alsa-lib \
openssl \
opus \
speex \
speexdsp
# Copy PJSIP from builder
COPY --from=builder /opt/pjsip /usr/local
COPY --from=builder /usr/lib/python3.*/site-packages/pjsua2* /usr/lib/python3.11/site-packages/
# Update library cache
RUN ldconfig /usr/local/lib || true
gcc \
musl-dev \
python3-dev
# Set working directory
WORKDIR /app

2
sip-notifier/config.yaml
View File

@@ -1,5 +1,5 @@
name: "SIP Voice Notifier"
version: "2.0.2"
version: "2.0.3"
slug: "sip-notifier"
description: "Send voice notifications via SIP phone calls (includes integration)"
arch:

1
sip-notifier/requirements.txt
View File

@@ -2,3 +2,4 @@ flask==3.0.0
requests==2.31.0
pydub==0.25.1
gtts==2.5.0
pyVoIP==1.6.6

188
sip-notifier/sip_service.py
View File

@@ -1,5 +1,5 @@
#!/usr/bin/env python3
"""SIP Voice Notifier Add-on Service."""
"""SIP Voice Notifier Add-on Service using pyVoIP."""
import json
import logging
import os
@@ -10,7 +10,7 @@ from urllib.parse import urlparse
import requests
from flask import Flask, request, jsonify
from pydub import AudioSegment
import pjsua2 as pj
from pyVoIP.VoIP import VoIPPhone, InvalidStateError, CallState
# Configure logging
logging.basicConfig(
@@ -26,80 +26,6 @@ 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)
@@ -167,89 +93,68 @@ def generate_tts_audio(message: str) -> str:
def place_sip_call(destination: str, audio_file: str, duration: int):
"""Place a SIP call."""
ep = pj.Endpoint()
ep.libCreate()
"""Place a SIP call using pyVoIP."""
sip_user = CONFIG['sip_user']
sip_server = CONFIG['sip_server']
sip_password = CONFIG['sip_password']
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")
_LOGGER.info(f"Connecting to SIP server: {sip_server}")
try:
acc = SIPAccount()
acc_cfg = pj.AccountConfig()
# Create VoIP phone
phone = VoIPPhone(
sip_server,
5060,
sip_user,
sip_password,
callCallback=None,
myIP=None,
sipPort=5060,
rtpPortLow=10000,
rtpPortHigh=20000
)
sip_user = CONFIG['sip_user']
sip_server = CONFIG['sip_server']
sip_password = CONFIG['sip_password']
phone.start()
time.sleep(2) # Wait for registration
if not sip_user.startswith('sip:'):
sip_uri = f"sip:{sip_user}@{sip_server}"
else:
sip_uri = sip_user
_LOGGER.info(f"Calling: {destination}")
acc_cfg.idUri = sip_uri
acc_cfg.regConfig.registrarUri = f"sip:{sip_server}"
# Make call
call = phone.call(destination)
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:
# Wait for call to be answered
timeout = 10
elapsed = 0
while call.state != CallState.ANSWERED and elapsed < timeout:
time.sleep(0.5)
wait_time += 0.5
elapsed += 0.5
if not call.connected:
_LOGGER.warning("Call did not connect")
if call.state != CallState.ANSWERED:
_LOGGER.warning("Call not answered within timeout")
call.hangup()
phone.stop()
return
_LOGGER.info("Call answered! Playing audio...")
# Transmit audio file
call.write_audio(audio_file)
# Keep call active
_LOGGER.info(f"Call active for {duration}s")
time.sleep(duration)
# Hangup
_LOGGER.info("Hanging up...")
hangup_param = pj.CallOpParam()
call.hangup(hangup_param)
call.hangup()
time.sleep(1)
# Cleanup
phone.stop()
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():
@@ -259,10 +164,7 @@ def health():
@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.
"""
"""Handle send_notification service call."""
try:
data = request.json
destination = data.get('destination')