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Commit a089b323 authored by Alexandre Savard's avatar Alexandre Savard
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[#4233] Add sdp related unit tests

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sflphone-common/test/sippxml/test_4.xml 0 → 100644
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View file @ a089b323
<?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>
sflphone-common/test/siptest.cpp
+ 43
− 0
View file @ a089b323
......@@ -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;
}
sflphone-common/test/siptest.h
+ 2
− 0
View file @ a089b323
......@@ -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:
};
......
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