diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index b22e053aca8f0f0d1805a5d1883d56de13b7e6d0..a7058bbb7bed2735fae67677c1a973fa6ff74cc6 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -2404,7 +2404,10 @@ int SIPVoIPLink::createUdpTransport (AccountID id)
 
     }
 
-
+    if(listeningAddress == "" || listeningPort == 0) {
+    	_error("UserAgent: Error invalid address for new udp transport");
+    	return !PJ_SUCCESS;
+    }
     //strcpy (tmpIP, listeningAddress.data());
     /* Init published name */
     pj_bzero (&a_name, sizeof (pjsip_host_port));
diff --git a/sippxml/accountcalluac.xml b/sippxml/accountcalluac.xml
new file mode 100644
index 0000000000000000000000000000000000000000..2d43c03840e0f721f8abda4f3d7df9217d5fe0e9
--- /dev/null
+++ b/sippxml/accountcalluac.xml
@@ -0,0 +1,161 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or      -->
+<!-- modify it under the terms of the GNU General Public License as     -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version.                    -->
+<!--                                                                    -->
+<!-- This program is distributed in the hope that it will be useful,    -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
+<!-- GNU General Public License for more details.                       -->
+<!--                                                                    -->
+<!-- You should have received a copy of the GNU General Public License  -->
+<!-- along with this program; if not, write to the                      -->
+<!-- Free Software Foundation, Inc.,                                    -->
+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
+
+
+<scenario name="accountcall_client">
+
+  <send retrans="500">
+    <![CDATA[
+
+      REGISTER sip:192.168.50.79 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+      Max-Forward: 70
+      From: <sip:27182@[local_ip]:[local_port]>;tag=[call_number]
+      To: <sip:27182@192.168.50.79:[local_port]>
+      Call-ID: [call_id]
+      CSeq: 1 REGISTER
+      Contact: <sip:27182@[local_ip]:[local_port]>
+      Content-Length: 0
+      Expires: 300
+
+    ]]>
+  </send>
+
+  <recv response="401" auth="true">
+  </recv>
+
+  <send retrans="500">
+    <![CDATA[
+
+      REGISTER sip:192.168.50.79 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+      Max-Forwards: 70
+      From: <sip:27182@[local_ip]:[local_port]>;tag=[call_number]
+      To: <sip:27182@192.168.50.79:[local_port]>
+      Call-ID: [call_id]
+      CSeq: 1 REGISTER
+      Contact: <sip:27182@[local_ip]:[local_port]>
+      Content-Length: 0
+      Expires: 300
+      [authentication username=27182 password=1234]
+
+    ]]>
+  </send>
+
+  <recv response="200">
+  </recv>
+
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:31416@192.168.50.79 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:27182@[local_ip]:[local_port]>;tag=[call_number]
+      To: <sip:31416@192.168.50.79:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:27182@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Functional Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+
+  <!-- Do something different on an optional receive   -->
+  <recv response="403" optional="true" next="1">
+  </recv>
+
+  <recv response="200">
+    <action>
+      <ereg regexp="ua25"
+            search_in="hdr"
+            header="From: "
+            assign_to="8"/>
+    </action>
+  </recv>
+
+  <!-- set variable 8 above on 25th call, send the ACK but skip the pause for it   -->
+  <send next="1" test="8">
+    <![CDATA[
+
+      ACK sip:31416@192.168.50.79 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:27182@[local_ip]:[local_port]>;tag=[call_number]
+      To: <sip:31416@192.168.50.79:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:27182@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Functional Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="5000"/>
+
+  <label id="1"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:31416@192.168.50.79 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:27182@[local_ip]:[local_port]>;tag=[call_number]
+      To: <sip:31416@192.168.50.79:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:27182@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Functional Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <pause milliseconds="4000"/>
+
+
+  <!-- definition of the response time repartition table (unit is ms)   -->
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <!-- definition of the call length repartition table (unit is ms)     -->
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
diff --git a/sippxml/accountcalluas.xml b/sippxml/accountcalluas.xml
new file mode 100644
index 0000000000000000000000000000000000000000..82158ef0a5c0fd85053aa9232d387d55e2ed219c
--- /dev/null
+++ b/sippxml/accountcalluas.xml
@@ -0,0 +1,159 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or      -->
+<!-- modify it under the terms of the GNU General Public License as     -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version.                    -->
+<!--                                                                    -->
+<!-- This program is distributed in the hope that it will be useful,    -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
+<!-- GNU General Public License for more details.                       -->
+<!--                                                                    -->
+<!-- You should have received a copy of the GNU General Public License  -->
+<!-- along with this program; if not, write to the                      -->
+<!-- Free Software Foundation, Inc.,                                    -->
+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
+
+
+<scenario name="branch_server">
+  <recv request="REGISTER">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+      Expires: 300
+
+    ]]>
+  </send>
+
+  <!-- Set variable 3 if the ua is of the form ua2... -->
+  <recv request="INVITE" crlf="true">
+    <action>
+      <ereg regexp="ua2"
+            search_in="hdr"
+            header="From: "
+            assign_to="3"/>
+    </action>
+  </recv>
+
+  <!-- send 180 then trying if variable 3 is set -->
+  <send next="1" test="3">
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- if not, send a 403 error then skip to wait for a BYE -->
+  <send next="2">
+    <![CDATA[
+
+      SIP/2.0 403 Error
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <label id="1"/>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 100 Trying
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Type: application/sdp
+      Content-Length: 136
+
+      v=0
+      o=user1 53655765 2353687637 IN IP4 127.0.0.1
+      s=-
+      t=0 0
+      c=IN IP4 [media_ip]
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK"
+        optional="true"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <label id="2"/>
+
+  <recv request="BYE">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Keep the call open for a while in case the 200 is lost to be     -->
+  <!-- able to retransmit it if we receive the BYE again.               -->
+  <pause milliseconds="4000"/>
+
+  <!-- Definition of the response time repartition table (unit is ms)   -->
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <!-- Definition of the call length repartition table (unit is ms)     -->
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
diff --git a/sippxml/simpleuac.xml b/sippxml/ip2ipcalluac.xml
similarity index 100%
rename from sippxml/simpleuac.xml
rename to sippxml/ip2ipcalluac.xml
diff --git a/sippxml/simpleuas.xml b/sippxml/ip2ipcalluas.xml
similarity index 100%
rename from sippxml/simpleuas.xml
rename to sippxml/ip2ipcalluas.xml
diff --git a/sippxml/testsuiteuac.sh b/sippxml/testsuiteuac.sh
index 9188b130ce8e2f86f45dddeba9f82ba6d7516384..9864f1303c107ba60cb723e4d4b4a1ff9f653baf 100644
--- a/sippxml/testsuiteuac.sh
+++ b/sippxml/testsuiteuac.sh
@@ -3,25 +3,39 @@
 
 SERVERPORT=5062
 
-#start sipp server to receive calls from sflphone
-sipp -sf simpleuas.xml -p ${SERVERPORT} &
+function test_ip2ipcall {
 
-#start sflphoned
-/usr/lib/sflphone/sflphoned& 
+    #start sipp server to receive calls from sflphone
+    sipp -sf ip2ipcalluas.xml -p ${SERVERPORT} &
 
-#wait some time to make sure sflphone is started
-sleep 2;
+    #start sflphoned
+    /usr/lib/sflphone/sflphoned& 
 
-#run python client and associated script
-python ../tools/pysflphone/pysflphone_testdbus.py
+    #wait some time to make sure sflphone is started
+    sleep 2;
 
-#kill every one
-killall sipp
-killall sflphoned
+    #run python client and associated script
+    python ../tools/pysflphone/pysflphone_testdbus.py
+
+    #kill every one
+    killall sipp
+    killall sflphoned
+}
+
+function test_accountcall {
+
+    sipp -sf accountcalluac.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT}
+
+}
 
 # function called if CTRL-C detected
 bashtrap()
 {
     killall sipp
     killall sflphoned
-}
\ No newline at end of file
+}
+
+
+# Here Start the Test suite
+
+test_ip2ipcall
\ No newline at end of file
diff --git a/tools/pysflphone/pysflphone_testdbus.py b/tools/pysflphone/pysflphone_testdbus.py
index bf2598881455dbaf2e9cbda066374f31b5d40843..47183d6454dcc0221b3ae5a9eeaa27d1564d6214 100644
--- a/tools/pysflphone/pysflphone_testdbus.py
+++ b/tools/pysflphone/pysflphone_testdbus.py
@@ -9,6 +9,7 @@ from sflphonectrlsimple import SflPhoneCtrlSimple
 
 class SflPhoneTests(SflPhoneCtrlSimple):
 
+
     def test_get_allaccounts_methods(self):
 
         for account in self.getAllAccounts():
@@ -55,7 +56,7 @@ class SflPhoneTests(SflPhoneCtrlSimple):
 
 
     def test_create_account(self):
-        """Create a new sip fake account and remove it"""
+        """Create a new sip account"""
 
         CONFIG_ACCOUNT_TYPE = "Account.type"  
 	CONFIG_ACCOUNT_ALIAS = "Account.alias"
@@ -63,25 +64,34 @@ class SflPhoneTests(SflPhoneCtrlSimple):
 	USERNAME = "username"
 	PASSWORD = "password"
 	
-        accDetails = {CONFIG_ACCOUNT_TYPE:"SIP", CONFIG_ACCOUNT_ALIAS:"myNewAccount",
-                      HOSTNAME:"192.168.50.3", USERNAME:"431",
-                      PASSWORD:"alexandre"}
+        accDetails = {CONFIG_ACCOUNT_TYPE:"SIP", CONFIG_ACCOUNT_ALIAS:"testsuiteaccount",
+                      HOSTNAME:"192.168.50.79", USERNAME:"31416",
+                      PASSWORD:"1234"}
 
 
         accountID = self.addAccount(accDetails)
         print "New Account ID " + accountID
-        time.sleep(3)
+
+        return accountID
+
+
+    def test_remove_account(self, accountID):
+        """Remove test account"""
 
         self.removeAccount(accountID)
         print "Account with ID " + accountID + " removed"
 
 
+# Open sflphone and connect to sflphoned through dbus 
 sflphone = SflPhoneTests()
 
-# sflphone.test_get_allaccounts_methods()
+# Test 1: Makke approximately one IP2IP call per second 
+# to a sipp uas on local addrress
+#sflphone.test_make_iptoip_call()
 
-sflphone.test_make_iptoip_call()
 
+# Test 2: 
+accountID = sflphone.test_create_account()
 # sflphone.test_make_account_call()
-
-# sflphone.test_create_account()
+time.sleep(0.3)
+# sflphone.test_remove_account(accountID)