Merge pull request tayler6000#39 from tayler6000/bugfix/issue_27

Fixed issue tayler6000#27 and tayler6000#38

docs/VoIP.rst

@@ -15,6 +15,9 @@ There are two errors under ``pyVoIP.VoIP``.
 
 *exception* VoIP.\ **InvalidRangeError**
   This is thrown by :ref:`VoIPPhone` when you define the rtpPort ranges as rtpPortLow > rtpPortHigh.  However this is not checked by :ref:`VoIPCall`, so if you are using your own class instead of VoIPPhone, make sure these ranges are correct.
+
+*exception* VoIP.\ **NoPortsAvailableError**
+  This is thrown when a call is attempting to be initiated but no ports are available.
 
 Enums
 ***********
@@ -52,16 +55,19 @@ VoIPCall
 
 The VoIPCall class is used to represent a single VoIP Session, which may be to multiple :term:`clients<client>`.
 
-*class* VoIP.\ **VoIPCall**\ (phone, request, session_id, myIP, rtpPortLow, rtpPortHigh)
+*class* VoIP.\ **VoIPCall**\ (phone, callstate, request, session_id, myIP, ms)
       The *phone* argument is the initating instance of :ref:`VoIPPhone`.
 
+      The *callstate* arguement is the initiating :ref:`CallState<callstate>`.
+
       The *request* argument is the :ref:`SIPMessage` representation of the SIP INVITE request from the VoIP server.
 
       The *session_id* argument is a unique code used to identify the session with `SDP <https://tools.ietf.org/html/rfc4566#section-5.2>`_ when answering the call.
 
       The *myIP* argument is the IP address it will pass to :ref:`RTPClient`'s to bind to.
 
-      The *rtpPortLow* and *rtpPortHigh* arguments are used to generate random ports to use for audio transfer.  Per RFC 4566 Sections `5.7 <https://tools.ietf.org/html/rfc4566#section-5.7>`_ and `5.14 <https://tools.ietf.org/html/rfc4566#section-5.14>`_, it can take multiple ports to fully communicate with other :term:`clients<client>`, as such a large range is recommended.
+      The *ms* arguement is a dictionary with int as the key and a :ref:`PayloadType<payload-type>` as the value.  This is only used when originating the call.
+
 
     **dtmfCallback**\ (code)
       This method is called by :ref:`RTPClient`'s when a telephone-event DTMF message is received.  The *code* argument is a string.  It should be an Event in complinace with `RFC 4733 Section 3.2 <https://tools.ietf.org/html/rfc4733#section-3.2>`_.
@@ -126,10 +132,16 @@ The VoIPPhone class is used to manage the :ref:`SIPClient` class and create :ref
     The *rtpPortLow* and *rtpPortHigh* arguments are used to generate random ports to use for audio transfer.  Per RFC 4566 Sections `5.7 <https://tools.ietf.org/html/rfc4566#section-5.7>`_ and `5.14 <https://tools.ietf.org/html/rfc4566#section-5.14>`_, it can take multiple ports to fully communicate with other :term:`clients<client>`, as such a large range is recommended.  If an invalid range is given, a :ref:`InvalidStateError<invalidstateerror>` will be thrown.
 
   **callback**\ (request)
-    This method is called by the :ref:`SIPClient` when an INVITE or BYE request is received.  This function then creates a :ref:`VoIPCall` or terminates it respectively.  When a VoIPCall is created, it will then pass it to the *callCallback* function as an argument.  If *callCallback* is set to None, this function replies as BUSY. **This function should not be called by the** :term:`user`.
+    This method is called by the :ref:`SIPClient` when an INVITE or BYE request is received.  This method then creates a :ref:`VoIPCall` or terminates it respectively.  When a VoIPCall is created, it will then pass it to the *callCallback* function as an argument.  If *callCallback* is set to None, this function replies as BUSY. **This function should not be called by the** :term:`user`.
+
+  **request_port**\ (blocking=True)
+    This method is called when a new port is needed to use in a :ref:`VoIPCall`.  If blocking is set to True, this will wait until a port is available.  Otherwise, it will raise NoPortsAvailableError.
+
+  **release_ports**\ (call=None)
+    This method is called when a call ends.  If call is provided, it will only release the ports used by that :ref:`VoIPCall`.  Otherwise, it will iterate through all active calls, and release all ports that are no longer in use.
 
   **start**\ ()
-    This method starts the :ref:`SIPClient` class.
+    This method starts the :ref:`SIPClient` class.  On failure, this will automatically call stop().
 
   **stop**\ ()
     This method ends all currently ongoing calls, then stops the :ref:`SIPClient` class

docs/conf.py

@@ -24,7 +24,7 @@
 author = 'Tayler J Porter'
 
 # The full version, including alpha/beta/rc tags
-release = '1.5.4'
+release = '1.5.5'
 
 master_doc = 'index'
 

pyVoIP/SIP.py

@@ -570,7 +570,7 @@ def parseMessage(self, message):
 
   def start(self):
     if self.NSD == True:
-      raise RunTimeError("Attempted to start already started SIPClient")
+      raise RuntimeError("Attempted to start already started SIPClient")
     self.NSD = True
     self.s = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
     #self.out = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
@@ -583,16 +583,22 @@ def start(self):
 
   def stop(self):
     self.NSD = False
-    self.registerThread.cancel()
-    self.deregister()
-    self.s.close()
-    self.out.close()
+    if self.registerThread:
+      self.registerThread.cancel()
+      self.deregister()
+    self._close_sockets()
+
+  def _close_sockets(self):
+    if hasattr(self, 's') and self.s:
+        self.s.close()
+    if hasattr(self, 'out') and self.out:
+        self.out.close()
 
   def genCallID(self):
     return hashlib.sha256(str(self.callID.next()).encode('utf8')).hexdigest()[0:32]+"@"+self.myIP+":"+str(self.myPort)
 
   def genTag(self):
-    while True:
+    while True:  # Keep as True instead of NSD so it can generate a tag on deregister.
       tag = hashlib.md5(str(random.randint(1, 4294967296)).encode('utf8')).hexdigest()[0:8]
       if tag not in self.tags:
         self.tags.append(tag)

pyVoIP/VoIP.py

@@ -1,13 +1,15 @@
 from enum import Enum
 from pyVoIP import SIP, RTP
 from threading import Timer, Lock
+from typing import Optional
 import io
 import pyVoIP
 import random
 import socket
+import time
 import warnings
 
-__all__ = ['CallState', 'InvalidRangeError', 'InvalidStateError', 'VoIPCall', 'VoIPPhone']
+__all__ = ['CallState', 'InvalidRangeError', 'InvalidStateError', 'NoPortsAvailableError', 'VoIPCall', 'VoIPPhone']
 debug = pyVoIP.debug
 
 class InvalidRangeError(Exception):
@@ -16,6 +18,9 @@ class InvalidRangeError(Exception):
 class InvalidStateError(Exception):
   pass
 
+class NoPortsAvailableError(Exception):
+  pass
+
 class CallState(Enum):
   DIALING = "DIALING"
   RINGING = "RINGING"
@@ -26,16 +31,16 @@ class CallState(Enum):
 For initiating a phone call, try sending the packet and the recieved OK packet will be sent to the VoIPCall request header.
 '''
 class VoIPCall():
-  def __init__(self, phone, callstate, request, session_id, myIP, portRange=(10000, 20000), ms = None):
+  def __init__(self, phone, callstate, request, session_id, myIP, ms = None):
     self.state = callstate
     self.phone = phone
     self.sip = self.phone.sip
     self.request = request
     self.call_id = request.headers['Call-ID']
     self.session_id = str(session_id)
     self.myIP = myIP
-    self.rtpPortHigh = portRange[1]
-    self.rtpPortLow = portRange[0]
+    self.rtpPortHigh = self.phone.rtpPortHigh
+    self.rtpPortLow = self.phone.rtpPortLow
 
     self.dtmfLock = Lock()
     self.dtmf = io.StringIO()
@@ -107,13 +112,7 @@ def __init__(self, phone, callstate, request, session_id, myIP, portRange=(10000
             codecs[m] = assoc[m]
         #TODO: If no codecs are compatible then send error to PBX.
 
-        port = None
-        while port == None:
-          proposed = random.randint(self.rtpPortLow, self.rtpPortHigh)
-          if not proposed in self.phone.assignedPorts:
-            self.phone.assignedPorts.append(proposed)
-            self.assignedPorts[proposed] = codecs
-            port = proposed
+        port = self.phone.request_port()
         for ii in range(len(request.body['c'])):
           self.RTPClients.append(RTP.RTPClient(codecs, self.myIP, port, request.body['c'][ii]['address'], i['port']+ii, request.body['a']['transmit_type'], dtmf=self.dtmfCallback)) #TODO: Check IPv4/IPv6
     elif callstate == CallState.DIALING:
@@ -122,6 +121,10 @@ def __init__(self, phone, callstate, request, session_id, myIP, portRange=(10000
         self.port = m
         self.assignedPorts[m] = self.ms[m]
 
+  def __del__(self):
+    if hasattr(self, 'phone'):
+      self.phone.release_ports(call=self)
+
   def dtmfCallback(self, code):
     self.dtmfLock.acquire()
     bufferloc = self.dtmf.tell()
@@ -223,8 +226,10 @@ def deny(self):
       raise InvalidStateError("Call is not ringing")
     message = self.sip.genBusy(self.request)
     self.sip.out.sendto(message.encode('utf8'), (self.phone.server, self.phone.port))
-    self.RTPClients = []
+    for x in self.RTPClients:
+      x.stop()
     self.state = CallState.ENDED
+    del self.phone.calls[self.request.headers['Call-ID']]
 
   def hangup(self):
     if self.state != CallState.ANSWERED:
@@ -265,7 +270,9 @@ def __init__(self, server, port, username, password, callCallback=None, myIP=Non
 
     self.rtpPortLow = rtpPortLow
     self.rtpPortHigh = rtpPortHigh
+    self.NSD = False
 
+    self.portsLock = Lock()
     self.assignedPorts = []
     self.session_ids = []
 
@@ -280,6 +287,8 @@ def __init__(self, server, port, username, password, callCallback=None, myIP=Non
     self.callCallback = callCallback
 
     self.calls = {}    
+    self.threads = []
+    self.threadLookup = {}  # Allows you to find call ID based of thread.
     self.sip = SIP.SIPClient(server, port, username, password, myIP=self.myIP, myPort=sipPort, callCallback=self.callback)
 
   def callback(self, request):
@@ -307,15 +316,17 @@ def callback(self, request):
               sess_id = proposed
           message = self.sip.genRinging(request)
           self.sip.out.sendto(message.encode('utf8'), (self.server, self.port))
-          self.calls[call_id] = VoIPCall(self, CallState.RINGING, request, sess_id, self.myIP, portRange=(self.rtpPortLow, self.rtpPortHigh))
+          self.calls[call_id] = VoIPCall(self, CallState.RINGING, request, sess_id, self.myIP))
           try:
             t = Timer(1, self.callCallback, [self.calls[call_id]])
             t.name = "Phone Call: "+call_id
             t.start()
-          except Exception as e:
+            self.threads.append(t)
+            self.threadLookup[t] = call_id
+          except Exception:
             message = self.sip.genBusy(request)
             self.sip.out.sendto(message.encode('utf8'), (self.server, self.port))
-            raise e
+            raise
       elif request.method == "BYE":
         if not call_id in self.calls:
           return
@@ -348,12 +359,72 @@ def callback(self, request):
         debug("Terminating Call")
         ack = self.sip.genAck(request)
         self.sip.out.sendto(ack.encode('utf8'), (self.server, self.port))
-
+
+  def request_port(self, blocking=True) -> int:
+    ports_available = [port for port in range(self.rtpPortLow,
+                       self.rtpPortHigh + 1) if port not in self.assignedPorts]
+    if len(ports_available) == 0:
+      # If no ports are available attempt to cleanup any missed calls.
+      self.release_ports()
+      ports_available = [port for port in range(self.rtpPortLow,
+                         self.rtpPortHigh + 1) if (port not in
+                                                   self.assignedPorts)]
+
+    while self.NSD and blocking and len(ports_available) == 0:
+      ports_available = [port for port in range(self.rtpPortLow,
+                         self.rtpPortHigh + 1) if (port not in
+                                                   self.assignedPorts)]
+      time.sleep(.5)
+      self.release_ports()
+
+    if len(ports_available) == 0:
+        raise NoPortsAvailableError("No ports were available to be assigned")
+
+    selection = random.choice(ports_available)
+    self.assignedPorts.append(selection)
+
+    return selection
+
+  def release_ports(self, call: Optional[VoIPCall] = None) -> None:
+    self.portsLock.acquire()
+    self._cleanup_dead_calls()
+    try:
+      if isinstance(call, VoIPCall):
+        ports = list(call.assignedPorts.keys())
+      else:
+        dnr_ports = []
+        for call_id in self.calls:
+          dnr_ports += list(self.calls[call_id].assignedPorts.keys())
+        ports = []
+        for port in self.assignedPorts:
+            if port not in dnr_ports:
+                ports.append(port)
+
+      for port in ports:
+        index = self.assignedPorts.index(port)
+        self.assignedPorts.pop(index)
+    finally:
+      self.portsLock.release()
+
+  def _cleanup_dead_calls(self) -> None:
+    to_delete =[]
+    for thread in self.threads:
+      if not thread.is_alive():
+        call_id = self.threadLookup[thread]
+        del self.calls[call_id]
+        del self.threadLookup[thread]
+        to_delete.append(thread)
+    for thread in to_delete:
+      index = self.threads.index(thread)
+      self.threads.pop(thread)
+
   def start(self):
     try:
       self.sip.start()
-    except Exception:
+      self.NSD = True
+    except BaseException:
       self.sip.stop()
+      self.NSD = False
       raise
 
   def stop(self):
@@ -365,12 +436,7 @@ def stop(self):
     self.sip.stop()
 
   def call(self, number):
-    port = None
-    while port == None:
-      proposed = random.randint(self.rtpPortLow, self.rtpPortHigh)
-      if not proposed in self.assignedPorts:
-        self.assignedPorts.append(proposed)
-        port = proposed
+    port = self.request_port()
     medias = {}
     medias[port] = {0: pyVoIP.RTP.PayloadType.PCMU, 101: pyVoIP.RTP.PayloadType.EVENT}
     request, call_id, sess_id = self.sip.invite(number, medias, pyVoIP.RTP.TransmitType.SENDRECV)

pyVoIP/__init__.py

@@ -1,7 +1,7 @@
 
 __all__ = ['SIP', 'RTP', 'VoIP']
 
-version_info = (1, 5, 4)
+version_info = (1, 5, 5)
 
 __version__ = ".".join([str(x) for x in version_info])
 

setup.py

@@ -7,7 +7,7 @@
 
 setup(
     name='pyVoIP',
-    version='1.5.4',
+    version='1.5.5',
     description='PyVoIP is a pure python VoIP/SIP/RTP library.',
     long_description=long_description,
     long_description_content_type="text/markdown",