diff --git a/sflphone-common/test/sippxml/test_2.xml b/sflphone-common/test/sippxml/test_2.xml
new file mode 100644
index 0000000000000000000000000000000000000000..003b0adac85c72ddfd91c7eac08fe39389037cc8
--- /dev/null
+++ b/sflphone-common/test/sippxml/test_2.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 -->
+<!-- -->
+<!-- Sipp default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Two simultaneous outgoing call 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">
+ </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>
+
+ <!-- Here is where the call is put on hold -->
+ <recv request="INVITE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[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=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ 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 request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <!-- 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 2a32a84c86ad867b7cc7f5003c172d2a50e8c6d0..fed8c7fec5addf831e95cf1d1d51b054a675d7bf 100644
--- a/sflphone-common/test/siptest.cpp
+++ b/sflphone-common/test/siptest.cpp
@@ -254,13 +254,18 @@ void SIPTest::testTwoIncomingIpCall ()
pthread_t firstCallThread, secondCallThread;
void *status;
+ pthread_attr_t attr;
+
+ pthread_attr_init(&attr);
+ pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
+
// the first call is supposed to be put on hold when answering teh second incoming call
std::string firstCallCommand("sipp -sf sippxml/test_2.xml 127.0.0.1 -i 127.0.0.1 -p 5064 -m 1");
// command to be executed by the thread, user agent client which initiate a call and hangup
std::string secondCallCommand("sipp -sn uac 127.0.0.1 -i 127.0.0.1 -p 5062 -m 1 -d 1000");
- int rc = pthread_create(&firstCallThread, NULL, sippThread, (void *)(&firstCallCommand));
+ int rc = pthread_create(&firstCallThread, &attr, sippThread, (void *)(&firstCallCommand));
if (rc) {
std::cout << "SIPTest: ERROR; return code from pthread_create()" << std::endl;
}
@@ -282,7 +287,7 @@ void SIPTest::testTwoIncomingIpCall ()
sleep(1);
- rc = pthread_create(&secondCallThread, NULL, sippThread, (void *)(&secondCallCommand));
+ rc = pthread_create(&secondCallThread, &attr, sippThread, (void *)(&secondCallCommand));
if(rc) {
std::cout << "SIPTest: Error; return code from pthread_create()" << std::endl;
}
@@ -313,7 +318,5 @@ void SIPTest::testTwoIncomingIpCall ()
std::cout << "SIPTest: completed join with thread 2" << std::endl;
-
- std::cout << "---------------------------" << std::endl;
}
diff --git a/sflphone-common/test/siptest.h b/sflphone-common/test/siptest.h
index b1da9e762e5d0effab2b956448133f7a3348eb08..7f0c94f0f1e92d10c2d437cc237ecc396a13db9b 100644
--- a/sflphone-common/test/siptest.h
+++ b/sflphone-common/test/siptest.h
@@ -53,10 +53,10 @@ class SIPTest : public CppUnit::TestCase {
* Use cppunit library macros to add unit test the factory
*/
CPPUNIT_TEST_SUITE( SIPTest );
- // CPPUNIT_TEST ( testSimpleOutgoingIpCall );
- // CPPUNIT_TEST ( testSimpleIncomingIpCall );
- // CPPUNIT_TEST ( testTwoOutgoingIpCall );
- CPPUNIT_TEST ( testTwoIncomingIpCall );
+ CPPUNIT_TEST ( testSimpleOutgoingIpCall );
+ CPPUNIT_TEST ( testSimpleIncomingIpCall );
+ CPPUNIT_TEST ( testTwoOutgoingIpCall );
+ // CPPUNIT_TEST ( testTwoIncomingIpCall );
CPPUNIT_TEST_SUITE_END();
public: