v1.5.0

NOTES

@@ -1,12 +1,14 @@
-New in v1.0.0:
-Originate Calls
-Added blocking feature to readAudio, this makes readAudio not return until there is data to be returned.  If blocking is off and data is not available, bytes(length) will be returned.
-Now properly generating SIP tags to comply with the RFC.
-Other bug fixes
+New in v1.5.0:
+Fixed bug where pyVoIP would accept all codecs proposed by the server even if not compatible.  Will now only accept PCMU, PCMA, and telephone-event.
+Added handling of Native Bridging tested with Asterisk 16 SIP re-invite (External RTP bridge), this seems to still have issues with Asterisk 18, but unsure if it's my hardphone.
+Changed the audio read function in RTP to return b'\x80'*length instead of bytes(length), doing so stops the popping on the client side when no audio is being written.
+Fixed issue with ending phone calls originated by user.
+Added handling of 404 Not Found and 503 Service Unavailable errors.
+Added compatiblity with Asterisk PJSIP.
+Fixed bug with multithreaded calling.
 
 Currently Known Issues:
-BYE request on originated calls causes a 500 Error on Asterisk 13 (other versions not tested).  Unsure what causes this, reach out if you have a fix.
-Currently does not work with PJSIP (Only tested with Asterisk 18)
+Some issues with bridiging with Asterisk 18, and possible other versions.  Bridging is not supported by all phones so it's unclear if it's supported by the softphone and hardphone I use to do my tests.
 
 Upcoming patches/changes:
-Adjust code to be compatible with Asterisk PJSIP.
+Add support for CANCEL requests.

README.md

@@ -1,9 +1,9 @@
 # pyVoIP
 PyVoIP is a pure python VoIP/SIP/RTP library.  Currently, it supports PCMA, PCMU, and telephone-event.
 
-Please note this is is still in development and can only originate calls with PCMU.  In future, it will be able to initiate calls in PCMA as well.
+Please note this is still in development.
 
-This library does not depend on a sound library, i.e. you can use any sound library that can handle linear sound data i.e. pyaudio or even wave.  Keep in mind PCMU only supports 8000Hz, 1 channel, 8 bit, audio.
+This library does not depend on a sound library, i.e. you can use any sound library that can handle linear sound data i.e. pyaudio or even wave.  Keep in mind PCMU/PCMA only supports 8000Hz, 1 channel, 8 bit, audio.
 
 ## Getting Started
 Simply put pyVoIP into your site-packages folder.

docs/Examples.rst

@@ -23,8 +23,8 @@ The following will create a phone that answers and automatically hangs up:
     except InvalidStateError:
       pass
   
-  if __name__=='__main__':
-    phone=VoIPPhone(<SIP Server IP>, <SIP Server Port>, <SIP Server Username>, <SIP Server Password>, callCallback=answer, myIP=<Your computer's local IP>)
+  if __name__ == '__main__':
+    phone = VoIPPhone(<SIP Server IP>, <SIP Server Port>, <SIP Server Username>, <SIP Server Password>, callCallback=answer, myIP=<Your computer's local IP>)
     phone.start()
     input('Press enter to disable the phone')
     phone.stop()
@@ -37,13 +37,12 @@ Let's say you want to make a phone that when you call it, it plays an announceme
 .. code-block:: python
 
   from pyVoIP.VoIP import VoIPPhone, InvalidStateError, CallState
-  import audioop
   import time
   import wave
   
   def answer(call):
     try:
-      f=wave.open('announcment.wav', 'rb')
+      f = wave.open('announcment.wav', 'rb')
       frames = f.getnframes()
       data = f.readframes(frames)
       f.close()
@@ -61,7 +60,7 @@ Let's say you want to make a phone that when you call it, it plays an announceme
     except:
       call.hangup()
       
-  if __name__=='__main__':
+  if __name__ == '__main__':
     phone=VoIPPhone(<SIP Server IP>, <SIP Server Port>, <SIP Server Username>, <SIP Server Password>, callCallback=answer, myIP=<Your computers local IP>)
     phone.start()
     input('Press enter to disable the phone')
@@ -88,13 +87,12 @@ We can use the above code to create `IVR Menus <https://en.wikipedia.org/wiki/In
 .. code-block:: python
 
   from pyVoIP.VoIP import VoIPPhone, InvalidStateError, CallState
-  import audioop
   import time
   import wave
   
   def answer(call):
     try:
-      f=wave.open('prompt.wav', 'rb')
+      f = wave.open('prompt.wav', 'rb')
       frames = f.getnframes()
       data = f.readframes(frames)
       f.close()
@@ -116,7 +114,7 @@ We can use the above code to create `IVR Menus <https://en.wikipedia.org/wiki/In
     except:
       call.hangup()
       
-  if __name__=='__main__':
+  if __name__ == '__main__':
     phone=VoIPPhone(<SIP Server IP>, <SIP Server Port>, <SIP Server Username>, <SIP Server Password>, callCallback=answer, myIP=<Your computers local IP>)
     phone.start()
     input('Press enter to disable the phone')

pyVoIP/RTP.py

@@ -9,6 +9,7 @@
 import time
 
 __all__ = ['add_bytes', 'byte_to_bits', 'DynamicPayloadType', 'PayloadType', 'RTPParseError', 'RTPProtocol', 'RTPPacketManager', 'RTPClient', 'TransmitType']
+debug = pyVoIP.debug
 
 def byte_to_bits(byte):
   byte = bin(ord(byte)).lstrip('-0b')
@@ -94,6 +95,7 @@ def __str__(self):
 
   #Non-codec
   EVENT = "telephone-event", 8000, 0, "telephone-event"
+  UNKOWN = "UNKOWN", 0, 0, "UNKOWN CODEC"
 
 class RTPPacketManager(): 
   def __init__(self):
@@ -109,7 +111,7 @@ def read(self, length=160):
     self.bufferLock.acquire()
     packet = self.buffer.read(length)
     if len(packet)<length:
-      packet = packet + bytes(length-len(packet))
+      packet = packet + (b'\x80' * (length-len(packet)))
     self.bufferLock.release()
     return packet
 
@@ -167,7 +169,7 @@ def parse(self, packet):
     byte = byte_to_bits(packet[0:1])
     self.version = int(byte[0:2], 2)
     if not self.version in self.RTPCompatibleVersions:
-      raise RTPParseError("RTP Version {} not compatible.".format(version))
+      raise RTPParseError("RTP Version {} not compatible.".format(self.version))
     self.padding = bool(int(byte[2], 2))
     self.extension = bool(int(byte[3], 2)) 
     self.CC = int(byte[4:], 2)
@@ -208,7 +210,16 @@ class RTPClient():
   def __init__(self, assoc, inIP, inPort, outIP, outPort, sendrecv, dtmf = None):
     self.NSD = True
     self.assoc = assoc # Example: {0: PayloadType.PCMU, 101: PayloadType.EVENT}
-    self.preference = PayloadType.PCMU
+    debug("Selecting audio codec for transmission")
+    for m in assoc:
+      try:
+        if int(assoc[m]) is not None:
+          debug(f"Selected {assoc[m]}")
+          self.preference = assoc[m] #Select the first available actual codec to encode with.  TODO: will need to change if video codecs are ever implemented.
+          break
+      except:
+        debug(f"{assoc[m]} cannot be selected as an audio codec")
+
     self.inIP = inIP
     self.inPort = inPort
     self.outIP = outIP
@@ -246,7 +257,7 @@ def read(self, length=160, blocking=True):
     if not blocking:
       return self.pmin.read(length)
     packet = self.pmin.read(length)
-    while packet == bytes(length) and self.NSD:
+    while packet == (b'\x80'*length) and self.NSD:
       time.sleep(0.01)
       packet = self.pmin.read(length)
     return packet
@@ -263,7 +274,7 @@ def recv(self):
       except BlockingIOError:
         time.sleep(0.01)
       except RTPParseError as e:
-        print(str(e))
+        debug(str(e))
       except OSError:
         pass
 
@@ -284,7 +295,7 @@ def trans(self):
       packet += self.outSSRC.to_bytes(4, byteorder='big')
       packet += payload
 
-      #print(payload)
+      #debug(payload)
 
       try:
         self.sout.sendto(packet, (self.outIP, self.outPort))

pyVoIP/SIP.py

@@ -8,6 +8,7 @@
 import time
 
 __all__ = ['Counter', 'InvalidAccountInfoError', 'SIPClient', 'SIPMessage', 'SIPMessageType', 'SIPParseError', 'SIPStatus']
+debug = pyVoIP.debug
 
 class InvalidAccountInfoError(Exception):
   pass
@@ -232,8 +233,12 @@ def parseHeader(self, header, data):
       contact = raw.split('<sip:')
       contact[0] = contact[0].strip('"').strip("'")
       address = contact[1].strip('>')
-      number = address.split('@')[0]
-      host = address.split('@')[1]
+      if len(address.split('@')) == 2:
+        number = address.split('@')[0]
+        host = address.split('@')[1]
+      else:
+        number = None
+        host = address
 
       self.headers[header] = {'raw': raw, 'tag': tag, 'address': address, 'number': number, 'caller': contact[0], 'host': host}
     elif header=="CSeq":
@@ -243,9 +248,11 @@ def parseHeader(self, header, data):
     elif header=="Content-Length":
       self.headers[header] = int(data)
     elif header=='WWW-Authenticate' or header=="Authorization":
-      info = data.split(", ")
+      data = data.replace("Digest", "")
+      info = data.split(",")
       header_data = {}
       for x in info:
+        x = x.strip()
         header_data[x.split('=')[0]] = x.split('=')[1].strip('"')
       self.headers[header] = header_data
       self.authentication = header_data
@@ -499,15 +506,15 @@ def __init__(self, server, port, username, password, myIP=None, myPort=5060, cal
 
     self.registerThread = None
     self.recvLock = Lock()
-
 
   def recv(self):
     while self.NSD:
       self.recvLock.acquire()
       self.s.setblocking(False)
       try:
-        message = SIPMessage(self.s.recv(8192))
-        #print(message.summary())
+        raw = self.s.recv(8192)
+        message = SIPMessage(raw)
+        debug(message.summary())
         self.parseMessage(message)
       except BlockingIOError:
         self.s.setblocking(True)
@@ -519,9 +526,13 @@ def recv(self):
           request = self.genSIPVersionNotSupported(message)
           self.out.sendto(request.encode('utf8'), (self.server, self.port))
         else:
-          print(str(e))
-      #except Exception as e:
-        #print("SIP recv error: "+str(e))
+          debug(f"SIPParseError in SIP.recv: {type(e)}, {e}")
+      except Exception as e:
+        debug(f"SIP.recv error: {type(e)}, {e}\n\n{str(raw, 'utf8')}")
+        if pyVoIP.DEBUG:
+          self.s.setblocking(True)
+          self.recvLock.release()
+          raise
       self.s.setblocking(True)
       self.recvLock.release()
 
@@ -530,17 +541,23 @@ def parseMessage(self, message):
       if message.status == SIPStatus.OK:
         if self.callCallback != None:
           self.callCallback(message)
+      elif message.status == SIPStatus.NOT_FOUND:
+        if self.callCallback != None:
+          self.callCallback(message)
+      elif message.status == SIPStatus.SERVICE_UNAVAILABLE:
+        if self.callCallback != None:
+          self.callCallback(message)
+      elif message.status == SIPStatus.TRYING or message.status == SIPStatus.RINGING:
+        pass
       else:
-        print("TODO: Add 500 Error on Receiving SIP Response")#:\r\n"+message.summary())
+        debug("TODO: Add 500 Error on Receiving SIP Response:\r\n"+message.summary(), "TODO: Add 500 Error on Receiving SIP Response")
       self.s.setblocking(True)
       return
     elif message.method == "INVITE":
       if self.callCallback == None:
         request = self.genBusy(message)
         self.out.sendto(request.encode('utf8'), (self.server, self.port))
       else:
-        request = self.genRinging(message)
-        self.out.sendto(request.encode('utf8'), (self.server, self.port))
         self.callCallback(message)
     elif message.method == "BYE":
       self.callCallback(message)
@@ -549,12 +566,13 @@ def parseMessage(self, message):
     elif message.method == "ACK":
       return
     else:
-      print("TODO: Add 400 Error on non processable request")
+      debug("TODO: Add 400 Error on non processable request")
 
   def start(self):
     self.s = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
-    self.out = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
+    #self.out = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
     self.s.bind((self.myIP, self.myPort))
+    self.out = self.s
     register = self.register()
     t = Timer(1, self.recv)
     t.name = "SIP Recieve"
@@ -668,7 +686,7 @@ def genRinging(self, request):
   def genAnswer(self, request, sess_id, ms, sendtype):
     #Generate body first for content length
     body = "v=0\r\n"
-    body += "o=pyVoIP "+sess_id+" "+sess_id+" IN IP4 "+self.myIP+"\r\n" #TODO: Check IPv4/IPv6
+    body += "o=pyVoIP "+sess_id+" "+str(int(sess_id)+2)+" IN IP4 "+self.myIP+"\r\n" #TODO: Check IPv4/IPv6
     body += "s=pyVoIP """+pyVoIP.__version__+"\r\n"
     body += "c=IN IP4 "+self.myIP+"\r\n" #TODO: Check IPv4/IPv6
     body += "t=0 0\r\n"
@@ -706,7 +724,7 @@ def genAnswer(self, request, sess_id, ms, sendtype):
   def genInvite(self, number, sess_id, ms, sendtype, branch, call_id):
     #Generate body first for content length
     body = "v=0\r\n"
-    body += "o=pyVoIP "+sess_id+" "+sess_id+" IN IP4 "+self.myIP+"\r\n" #TODO: Check IPv4/IPv6
+    body += "o=pyVoIP "+sess_id+" "+str(int(sess_id)+2)+" IN IP4 "+self.myIP+"\r\n" #TODO: Check IPv4/IPv6
     body += "s=pyVoIP """+pyVoIP.__version__+"\r\n"
     body += "c=IN IP4 "+self.myIP+"\r\n" #TODO: Check IPv4/IPv6
     body += "t=0 0\r\n"
@@ -754,7 +772,7 @@ def genBye(self, request):
       byeRequest += "To: "+request.headers['From']['raw']+";tag="+request.headers['From']['tag']+"\r\n"
       byeRequest += "From: "+request.headers['To']['raw']+";tag="+tag+"\r\n"
     byeRequest += "Call-ID: "+request.headers['Call-ID']+"\r\n"
-    byeRequest += "CSeq: "+str(self.byeCounter.next())+" BYE\r\n"
+    byeRequest += "CSeq: "+str(int(request.headers['CSeq']['check'])+1)+" BYE\r\n"
     byeRequest += "Contact: <sip:"+self.username+"@"+self.myIP+":"+str(self.myPort)+">\r\n"
     byeRequest += "User-Agent: pyVoIP """+pyVoIP.__version__+"\r\n"
     byeRequest += "Allow: "+(", ".join(pyVoIP.SIPCompatibleMethods))+"\r\n"
@@ -785,22 +803,21 @@ def invite(self, number, ms, sendtype):
     self.recvLock.acquire()
     #print("Locked")
     self.out.sendto(invite.encode('utf8'), (self.server, self.port))
-    #print('Invited')
+    debug('Invited')
     response = SIPMessage(self.s.recv(8192))
 
-    while response.status != SIPStatus(401) and response.status != SIPStatus(100) and response.status != SIPStatus(180):
+    while (response.status != SIPStatus(401) and response.status != SIPStatus(100) and response.status != SIPStatus(180)) or response.headers['Call-ID'] != call_id:
       if not self.NSD:
         break
-      if response.status == SIPStatus(100) or response.status == SIPStatus(180):
-        return SIPMessage(invite.encode('utf8')), call_id, sess_id
       self.parseMessage(response)
       response = SIPMessage(self.s.recv(8192))
 
-    #print("Received Response: "+response.summary())
-
+    if response.status == SIPStatus(100) or response.status == SIPStatus(180):
+      return SIPMessage(invite.encode('utf8')), call_id, sess_id
+    debug("Received Response: "+response.summary())
     ack = self.genAck(response)
     self.out.sendto(ack.encode('utf8'), (self.server, self.port))
-    #print("Acknowledged")
+    debug("Acknowledged")
     authhash = self.genAuthorization(response)
     nonce = response.authentication['nonce']
     auth = 'Authorization: Digest username="'+self.username
@@ -815,6 +832,7 @@ def invite(self, number, ms, sendtype):
     self.out.sendto(invite.encode('utf8'), (self.server, self.port))
 
     self.recvLock.release()
+    #print("Released")
     return SIPMessage(invite.encode('utf8')), call_id, sess_id
 
   def bye(self, request):
@@ -869,6 +887,9 @@ def register(self):
       else:
         self.parseMessage(response)
 
+    #debug(response.summary())
+    #debug(response.raw)
+
     regRequest = self.genRegister(response)
 
     self.out.sendto(regRequest.encode('utf8'), (self.server, self.port))