Replace PJSIP with pyVoIP - simpler, pure Python SIP library (v2.0.3)

sip-notifier/Dockerfile

@@ -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 \
+    gcc \
-    openssl \
+    musl-dev \
-    opus \
+    python3-dev
-    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
 
 # Set working directory
 WORKDIR /app

sip-notifier/config.yaml

@@ -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:

sip-notifier/requirements.txt

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

sip-notifier/sip_service.py

@@ -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."""
+    """Place a SIP call using pyVoIP."""
-    ep = pj.Endpoint()
+    sip_user = CONFIG['sip_user']
-    ep.libCreate()
+    sip_server = CONFIG['sip_server']
+    sip_password = CONFIG['sip_password']
     
-    ep_cfg = pj.EpConfig()
+    _LOGGER.info(f"Connecting to SIP server: {sip_server}")
-    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()
+        # Create VoIP phone
-        acc_cfg = pj.AccountConfig()
+        phone = VoIPPhone(
+            sip_server,
+            5060,
+            sip_user,
+            sip_password,
+            callCallback=None,
+            myIP=None,
+            sipPort=5060,
+            rtpPortLow=10000,
+            rtpPortHigh=20000
+        )
         
-        sip_user = CONFIG['sip_user']
+        phone.start()
-        sip_server = CONFIG['sip_server']
+        time.sleep(2)  # Wait for registration
-        sip_password = CONFIG['sip_password']
         
-        if not sip_user.startswith('sip:'):
+        _LOGGER.info(f"Calling: {destination}")
-            sip_uri = f"sip:{sip_user}@{sip_server}"
-        else:
-            sip_uri = sip_user
         
-        acc_cfg.idUri = sip_uri
+        # Make call
-        acc_cfg.regConfig.registrarUri = f"sip:{sip_server}"
+        call = phone.call(destination)
         
-        cred = pj.AuthCredInfo("digest", "*", sip_user, 0, sip_password)
+        # Wait for call to be answered
-        acc_cfg.sipConfig.authCreds.append(cred)
+        timeout = 10
-        
+        elapsed = 0
-        acc.create(acc_cfg)
+        while call.state != CallState.ANSWERED and elapsed < timeout:
-        _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
+            elapsed += 0.5
         
-        if not call.connected:
+        if call.state != CallState.ANSWERED:
-            _LOGGER.warning("Call did not connect")
+            _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()
-        call.hangup(hangup_param)
         
-        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."""
-    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')