diff --git a/sflphone-common/test/sippxml/test_4.xml b/sflphone-common/test/sippxml/test_4.xml
new file mode 100644
index 0000000000000000000000000000000000000000..bc4e5636fee7ad1998c5cd59747eed2edd3208c6
--- /dev/null
+++ b/sflphone-common/test/sippxml/test_4.xml
@@ -0,0 +1,122 @@
+<?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 -->
+<!-- -->
+<!-- Sipp default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Basic Sipstone UAC">
+ <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
+ <!-- generated by sipp. To do so, use [call_id] keyword. --
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance 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>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <!-- By adding rrs="true" (Record Route Sets), the route sets -->
+ <!-- are saved and used for following messages sent. Useful to test -->
+ <!-- against stateful SIP proxies/B2BUAs. -->
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="a=rtpmap:0 PCMU/8000" search_in="body" check_it="true" assign_to="1" />
+ <log message="Custom header is [$1]"/>
+ </action>
+ </recv>
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- This delay can be customized by the -d command-line option -->
+ <!-- or by adding a 'milliseconds = "value"' option here. -->
+ <pause/>
+
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <!-- 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/sflphone-common/test/siptest.cpp b/sflphone-common/test/siptest.cpp
index 27d15273578c70ffd35fc218af121be86e729c81..b764376c4283342fc06c158226974aebf1c1cdb7 100644
--- a/sflphone-common/test/siptest.cpp
+++ b/sflphone-common/test/siptest.cpp
@@ -408,3 +408,46 @@ void SIPTest::testHoldIpCall()
Manager::instance().hangupCall(testCallID);
}
+
+void SIPTest::testIncomingIpCallSdp ()
+{
+
+ pthread_t thethread;
+ void *status;
+
+ // command to be executed by the thread, user agent client which initiate a call and hangup
+ std::string command("sipp -sf sippxml/test_4.xml 127.0.0.1 -i 127.0.0.1 -p 5062 -m 1");
+
+ int rc = pthread_create(&thethread, NULL, sippThread, (void *)(&command));
+ if (rc) {
+ std::cout << "SIPTest: ERROR; return code from pthread_create()" << std::endl;
+ }
+
+
+ // sleep a while to make sure that sipp insdtance is initialized and sflphoned received
+ // the incoming invite.
+ sleep(2);
+
+ // gtrab call id from sipvoiplink
+ SIPVoIPLink *siplink = SIPVoIPLink::instance ("");
+
+ CPPUNIT_ASSERT(siplink->_callMap.size() == 1);
+ CallMap::iterator iterCallId = siplink->_callMap.begin();
+ std::string testcallid = iterCallId->first;
+
+ // TODO: hmmm, should IP2IP call be stored in call list....
+ CPPUNIT_ASSERT(Manager::instance().getCallList().size() == 0);
+
+ // Answer this call
+ CPPUNIT_ASSERT(Manager::instance().answerCall(testcallid));
+
+
+ sleep(1);
+
+ rc = pthread_join(thethread, &status);
+ if (rc) {
+ std::cout << "SIPTest: ERROR; return code from pthread_join(): " << rc << std::endl;
+ }
+ else
+ std::cout << "SIPTest: completed join with thread" << std::endl;
+}
diff --git a/sflphone-common/test/siptest.h b/sflphone-common/test/siptest.h
index 8810273bca1717ff0fc4b3cb48508a28a8e95f1a..abf5558f1536460e082b1c2f62f7362b527963d2 100644
--- a/sflphone-common/test/siptest.h
+++ b/sflphone-common/test/siptest.h
@@ -58,6 +58,7 @@ class SIPTest : public CppUnit::TestCase {
CPPUNIT_TEST ( testTwoOutgoingIpCall );
// CPPUNIT_TEST ( testTwoIncomingIpCall );
CPPUNIT_TEST ( testHoldIpCall);
+ CPPUNIT_TEST ( testIncomingIpCallSdp );
CPPUNIT_TEST_SUITE_END();
public:
@@ -86,6 +87,7 @@ class SIPTest : public CppUnit::TestCase {
void testHoldIpCall(void);
+ void testIncomingIpCallSdp(void);
private:
};