void testMultiplePAIs_Match()
{
const char* message =
"INVITE sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A29544 SIP/2.0\r\n"
"Record-Route: <sip:192.168.0.2:5060;lr>\r\n"
"From: caller <sip:[email protected]>;tag=94bc25b8-c0a80165-13c4-3e635-37aa1989-3e635\r\n"
"To: <sip:[email protected]>\r\n"
"Call-Id: 94bb2520-c0a80165-13c4-3e635-3ccd2971-3e635@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Max-Forwards: 19\r\n"
"Supported: replaces\r\n"
"Contact: <sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A14956>\r\n"
"P-Asserted-Identity: \"Some guy\" <sip:[email protected]>\r\n"
"P-Asserted-Identity: \"Some other guy\" <sip:[email protected]>\r\n"
"P-Asserted-Identity: \"yet another guy\" <sip:[email protected]>\r\n"
"P-Asserted-Identity: \"you again\" <sip:[email protected]>\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg(message, strlen(message));
UtlSList noRemovedRoutes;
UtlString routeName("mydomain.com");
RouteState routeState( testMsg, noRemovedRoutes, routeName );
UtlString rejectReason;
CPPUNIT_ASSERT(AuthPlugin::CONTINUE
== spLinter->authorizeAndModify(identity,
requestUri,
routeState,
"INVITE",
AuthPlugin::CONTINUE,
testMsg,
false,
rejectReason
));
// verify that PAIs are gone.
CPPUNIT_ASSERT( !testMsg.getHeaderValue( 0, SipXauthIdentity::PAssertedIdentityHeaderName ) );
}
void testPAIForMatchingDomain_StillSpiraling()
{
const char* message =
"INVITE sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A29544 SIP/2.0\r\n"
"Record-Route: <sip:192.168.0.2:5060;lr>\r\n"
"From: caller <sip:[email protected]>;tag=94bc25b8-c0a80165-13c4-3e635-37aa1989-3e635\r\n"
"To: <sip:[email protected]>\r\n"
"Call-Id: 94bb2520-c0a80165-13c4-3e635-3ccd2971-3e635@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Max-Forwards: 19\r\n"
"Supported: replaces\r\n"
"Contact: <sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A14956>\r\n"
"P-Asserted-Identity: \"Some guy\" <sip:[email protected]>\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg(message, strlen(message));
CPPUNIT_ASSERT( testMsg.getHeaderValue( 0, SipXauthIdentity::PAssertedIdentityHeaderName ) );
UtlSList noRemovedRoutes;
UtlString routeName("mydomain.com");
RouteState routeState( testMsg, noRemovedRoutes, routeName );
UtlString rejectReason;
CPPUNIT_ASSERT(AuthPlugin::CONTINUE
== spLinter->authorizeAndModify(identity,
requestUri,
routeState,
"INVITE",
AuthPlugin::CONTINUE,
testMsg,
true, // request still spiraling
rejectReason
));
// verify that PAI is still there because request is still spiraling.
const char* pPAI;
CPPUNIT_ASSERT( ( pPAI = testMsg.getHeaderValue( 0, SipXauthIdentity::PAssertedIdentityHeaderName ) ) );
ASSERT_STR_EQUAL( "\"Some guy\" <sip:[email protected]>", pPAI );
}
void testPAIForOtherDomain()
{
const char* message =
"INVITE sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A29544 SIP/2.0\r\n"
"Record-Route: <sip:192.168.0.2:5060;lr>\r\n"
"From: caller <sip:[email protected]>;tag=94bc25b8-c0a80165-13c4-3e635-37aa1989-3e635\r\n"
"To: <sip:[email protected]>\r\n"
"Call-Id: 94bb2520-c0a80165-13c4-3e635-3ccd2971-3e635@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Max-Forwards: 19\r\n"
"Supported: replaces\r\n"
"Contact: <sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A14956>\r\n"
"P-Asserted-Identity: \"Some guy\" <sip:[email protected]>\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg(message, strlen(message));
UtlSList noRemovedRoutes;
UtlString routeName("mydomain.com");
RouteState routeState( testMsg, noRemovedRoutes, routeName );
UtlString rejectReason;
CPPUNIT_ASSERT(AuthPlugin::CONTINUE
== spLinter->authorizeAndModify(identity,
requestUri,
routeState,
"INVITE",
AuthPlugin::CONTINUE,
testMsg,
false,
rejectReason
));
ssize_t size;
UtlString modifiedMessageText;
testMsg.getBytes( &modifiedMessageText, &size );
ASSERT_STR_EQUAL( message, modifiedMessageText.data() );
}
void stalledMediaStreamDetectionTest_StreamStalled()
{
SetFakeReceptionOfPacketsFlag( false );
RouteState::setSecret("fixed");
const char* message =
"INVITE sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A29544 SIP/2.0\r\n"
"Record-Route: <sip:192.168.0.2:5060;lr>\r\n"
"From: \"L3\"<sip:[email protected]>;tag=94bb9480-c0a80065-13c4-50045-20c55bd3-50045\r\n"
"To: <sip:[email protected]>\r\n"
"Call-Id: 94bb55a0-c0a80065-13c4-50045-61a3ad3-50045@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK-sipXecs-8130adee90e0277761fec4ee1c194a56838f~9d6590763d78764b93a42e94dbd6b75a\r\n"
"Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-50045-13890e4e-4f17a229\r\n"
"Max-Forwards: 19\r\n"
"Supported: replaces\r\n"
"User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A183385\r\n"
"Contact: <sip:[email protected]:5060;x-sipX-nonat>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: 301\r\n"
"Date: Fri, 13 Jun 2008 14:48:44 GMT\r\n"
"\r\n"
"\r\n"
"v=0\r\n"
"o=LGEIPP 14595 14595 IN IP4 192.168.0.101\r\n"
"s=SIP Call\r\n"
"c=IN IP4 192.168.0.101\r\n"
"t=0 0\r\n"
"m=audio 23016 RTP/AVP 18 4 0 8 101\r\n"
"a=rtpmap:18 G729/8000\r\n"
"a=rtpmap:4 G723/8000\r\n"
"a=rtpmap:0 PCMU/8000\r\n"
"a=rtpmap:8 PCMA/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
"a=fmtp:18 annexb=no\r\n"
"a=fmtp:101 0-11\r\n"
"a=sendrecv\r\n";
SipMessage testMsg(message, strlen(message));
UtlString rejectReason;
UtlSList emptyList;
RouteState routeState( message, emptyList, "192.168.0.2:5060" );
AuthPlugin::AuthResult result;
result = pNatTraversalAgent->authorizeAndModify( "id", //dummy
Url("sip:[email protected]:5060"), //dummy
routeState,
"INVITE",
AuthPlugin::CONTINUE,
testMsg,
false,
rejectReason );
// check auth result
CPPUNIT_ASSERT( result == AuthPlugin::CONTINUE );
routeState.update( &testMsg );
pNatTraversalAgent->handleOutputMessage( testMsg, "47.135.162.145", 29544 );
// Mow simulalate the reception of the response
const char* response =
"SIP/2.0 200 OK\r\n"
"From: \"L3\"<sip:[email protected]>;tag=94bb9480-c0a80065-13c4-50045-20c55bd3-50045\r\n"
"To: <sip:[email protected]>;tag=94bf81c0-c0a8010b-13c4-4fc07-650a9211-4fc07\r\n"
"Call-Id: 94bb55a0-c0a80065-13c4-50045-61a3ad3-50045@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-50045-13890e4e-4f17a229;id=1234-3\r\n"
"Supported: replaces\r\n"
"Contact: <sip:[email protected]:5060>\r\n"
"Record-Route: <sip:47.135.162.140:5060;lr;sipXecs-rs=%2Afrom%7EOTRiYjk0ODAtYzBhODAwNjUtMTNjNC01MDA0NS0yMGM1NWJkMy01MDA0NQ%60%60.ntap%2ACeT%7ENDcuMTM1LjE2Mi4xNDU6Mjk1NDQ7dHJhbnNwb3J0PXVkcA%60%60.ntap%2Aid%7EMTIzNC0z%213d7b53562948991d1cf59dfe7fdaaf24>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: 250\r\n"
"Date: Fri, 13 Jun 2008 14:48:46 GMT\r\n"
"\r\n"
"\r\n"
"v=0\r\n"
"o=LGEIPP 12617 12617 IN IP4 192.168.1.11\r\n"
"s=SIP Call\r\n"
"c=IN IP4 192.168.1.11\r\n"
"t=0 0\r\n"
"m=audio 25038 RTP/AVP 4 101\r\n"
"c=IN IP4 192.168.0.2\r\n"
"a=rtpmap:4 G723/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
"a=fmtp:101 0-11\r\n"
"a=sendrecv\r\n"
"a=x-sipx-ntap:192.168.0.2;3\r\n";
SipMessage okResp( response, strlen(response) );
pNatTraversalAgent->handleOutputMessage( okResp, "192.168.0.101", 5060 );
// Mow simulalate the reception of the ACK
const char* ack =
"ACK sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A29544 SIP/2.0\r\n"
"From: \"L3\"<sip:[email protected]>;tag=94bb9480-c0a80065-13c4-50045-20c55bd3-50045\r\n"
"To: <sip:[email protected]>;tag=94bf81c0-c0a8010b-13c4-4fc07-650a9211-4fc07\r\n"
"Call-Id: 94bb55a0-c0a80065-13c4-50045-61a3ad3-50045@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 ACK\r\n"
"Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK-sipXecs-8130adee90e0277761fec4ee1c194a56838f~9d6590763d78764b93a42e94dbd6b75a\r\n"
"Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-50045-13890e4e-4f17a229\r\n"
"Max-Forwards: 19\r\n"
"Supported: replaces\r\n"
"User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A183385\r\n"
"Contact: <sip:[email protected]:5060;x-sipX-nonat>\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage ackRequest( ack, strlen(ack) );
UtlSList routeList;
UtlString poppedRoute("<sip:47.135.162.140:5060;lr;sipXecs-rs=%2Afrom%7EOTRiYjk0ODAtYzBhODAwNjUtMTNjNC01MDA0NS0yMGM1NWJkMy01MDA0NQ%60%60.ntap%2ACeT%7ENDcuMTM1LjE2Mi4xNDU6Mjk1NDQ7dHJhbnNwb3J0PXVkcA%60%60.ntap%2Aid%7EMTIzNC0z%213d7b53562948991d1cf59dfe7fdaaf24>");
routeList.append( &poppedRoute );
RouteState newRouteState( message, routeList, "192.168.0.2:5060" );
result = pNatTraversalAgent->authorizeAndModify( "id", //dummy
Url("sip:[email protected]:5060"), //dummy
newRouteState,
"ACK",
AuthPlugin::CONTINUE,
ackRequest,
false,
rejectReason );
// check auth result
CPPUNIT_ASSERT( result == AuthPlugin::CONTINUE );
pNatTraversalAgent->handleOutputMessage( ackRequest, "47.135.162.145", 29544 );
// Check that the DialogTracker is still around after 10 seconds.
CallTracker* pTracker;
UtlString callId;
testMsg.getCallIdField( &callId );
pTracker = pNatTraversalAgent->getCallTrackerFromCallId( callId );
CPPUNIT_ASSERT( pTracker != 0 );
CPPUNIT_ASSERT( pTracker->mSessionContextsMap.entries() == 1 );
sleep( MAX_TIMER_TICK_COUNTS_BEFORE_CALL_TRACKER_CLEAN_UP * CLEAN_UP_TIMER_IN_SECS * 2 /*safety margin*/ );
pTracker = pNatTraversalAgent->getCallTrackerFromCallId( callId );
CPPUNIT_ASSERT( pTracker == 0 );
}
void handleRemoteNATedToRemoteNATedCall()
{
const char* message =
"INVITE sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A29544 SIP/2.0\r\n"
"Record-Route: <sip:192.168.0.2:5060;lr>\r\n"
"From: caller <sip:[email protected]>;tag=94bc25b8-c0a80165-13c4-3e635-37aa1989-3e635\r\n"
"To: <sip:[email protected]>\r\n"
"Call-Id: 94bb2520-c0a80165-13c4-3e635-3ccd2971-3e635@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Max-Forwards: 19\r\n"
"Supported: replaces\r\n"
"User-Agent: LG-Nortel LIP 6804 v1.2.38sp SN/00405A187376\r\n"
"Contact: <sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A14956>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: 297\r\n"
"Date: Thu, 12 Jun 2008 19:14:29 GMT\r\n"
"Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK-sipXecs-01a17f9662b96b1d454a71775bbda1ec6bf6~9d6590763d78764b93a42e94dbd6b75a\r\n"
"Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK-3e635-f3b41fc-310ddca7;received=47.135.162.145;rport=14956\r\n"
"\r\n"
"\r\n"
"v=0\r\n"
"o=LGEIPP 874 874 IN IP4 192.168.1.101\r\n"
"s=SIP Call\r\n"
"c=IN IP4 192.168.1.101\r\n"
"t=0 0\r\n"
"m=audio 23000 RTP/AVP 18 0 8 4 101\r\n"
"a=rtpmap:18 G729/8000\r\n"
"a=rtpmap:0 PCMU/8000\r\n"
"a=rtpmap:8 PCMA/8000\r\n"
"a=rtpmap:4 G723/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
"a=fmtp:18 annexb=no\r\n"
"a=fmtp:101 0-11\r\n"
"a=sendrecv\r\n";
SipMessage testMsg(message, strlen(message));
UtlString rejectReason;
UtlSList emptyList;
RouteState routeState( message, emptyList, "192.168.0.2:5060" );
AuthPlugin::AuthResult result;
result = pNatTraversalAgent->authorizeAndModify( "id", //dummy
Url("sip:[email protected]:5060"), //dummy
routeState,
"INVITE",
AuthPlugin::CONTINUE,
testMsg,
false,
rejectReason );
// check auth result
CPPUNIT_ASSERT( result == AuthPlugin::CONTINUE );
// check that call tracker got created
UtlString callId;
CallTracker* pTracker;
testMsg.getCallIdField( &callId );
pTracker = pNatTraversalAgent->getCallTrackerFromCallId( callId );
CPPUNIT_ASSERT( pTracker != 0 );
// check that public transport of caller got encoded in route state
UtlString transportInfo;
routeState.getParameter( pNatTraversalAgent->mInstanceName.data(), CALLER_PUBLIC_TRANSPORT_PARAM, transportInfo );
ASSERT_STR_EQUAL( "47.135.162.145:14956;transport=udp", transportInfo.data() );
// check that public transport of callee got encoded in route state
routeState.getParameter( pNatTraversalAgent->mInstanceName.data(), CALLEE_PUBLIC_TRANSPORT_PARAM, transportInfo );
ASSERT_STR_EQUAL( "47.135.162.145:29544;transport=udp", transportInfo.data() );
// Check that sipX NAT route is set to reach the callee's public IP address
UtlString natRoute;
CPPUNIT_ASSERT( testMsg.getSipXNatRoute( &natRoute ) == true );
ASSERT_STR_EQUAL( "47.135.162.145:29544;transport=udp", natRoute.data() );
// Now present the message to the Output Processor and verify that it got processed correctly.
routeState.update( &testMsg ); // uodate route state as it would normally be done by SipRouter.
pNatTraversalAgent->handleOutputMessage( testMsg, "47.135.162.145", 29544 );
// verify that top via got adjusted to reflect our public IP address
UtlString topmostVia;
CPPUNIT_ASSERT( testMsg.getViaFieldSubField( &topmostVia, 0 ) );
ASSERT_STR_EQUAL( "SIP/2.0/UDP 47.135.162.140:5060;branch=z9hG4bK-sipXecs-01a17f9662b96b1d454a71775bbda1ec6bf6~9d6590763d78764b93a42e94dbd6b75a",
topmostVia.data() );
// verify that top route got modfified to reflect our public address
UtlString tmpRecordRoute;
CPPUNIT_ASSERT( testMsg.getRecordRouteUri( 0, &tmpRecordRoute ) );
Url tmpRecordRouteUrl( tmpRecordRoute );
UtlString tmpRecordRouteHost;
tmpRecordRouteUrl.getHostWithPort( tmpRecordRouteHost );
ASSERT_STR_EQUAL( "47.135.162.140:5060", tmpRecordRouteHost.data() );
// recall the call tracker based on message and compare with previous
CallTracker* pTracker2;
pTracker2 = pNatTraversalAgent->getCallTrackerForMessage( testMsg );
CPPUNIT_ASSERT( pTracker2 == pTracker );
// check that the SDP got changed properly to relay media
const SdpBody* pSdpBody = testMsg.getSdpBody();
CPPUNIT_ASSERT( pSdpBody );
UtlString mediaType, mediaTransportType;
int mediaPort, mediaPortPairs, numPayloadTypes, payloadTypes[10];
SdpDirectionality dir;
CPPUNIT_ASSERT( pSdpBody->getMediaData( 0,
&mediaType,
&mediaPort,
&mediaPortPairs,
&mediaTransportType,
10,
&numPayloadTypes,
payloadTypes,
&dir ) == TRUE );
CPPUNIT_ASSERT( mediaType == "audio" );
CPPUNIT_ASSERT( mediaPort == 10000 );
CPPUNIT_ASSERT( mediaPortPairs == 1 );
CPPUNIT_ASSERT( mediaTransportType == "RTP/AVP" );
CPPUNIT_ASSERT( numPayloadTypes == 5 );
CPPUNIT_ASSERT( payloadTypes[0] == 18 );
CPPUNIT_ASSERT( payloadTypes[1] == 0 );
CPPUNIT_ASSERT( payloadTypes[2] == 8 );
CPPUNIT_ASSERT( payloadTypes[3] == 4 );
CPPUNIT_ASSERT( payloadTypes[4] == 101 );
CPPUNIT_ASSERT( dir == sdpDirectionalitySendRecv );
// check that we are using the public IP address of the media relay
UtlString mediaAddress;
CPPUNIT_ASSERT( pSdpBody->getMediaAddress(0, &mediaAddress ) );
ASSERT_STR_EQUAL( pNatTraversalAgent->mNatTraversalRules.getMediaRelayPublicAddress(), mediaAddress );
// Mow simulalate the reception of the response
const char* response =
"SIP/2.0 200 OK\r\n"
"From: \"R1_2 - 602\"<sip:[email protected]>;tag=94bc25b8-c0a80165-13c4-3e635-37aa1989-3e635\r\n"
"To: <sip:[email protected]>;tag=94bf84d0-c0a8010b-13c4-3e8d1-496994c2-3e8d1\r\n"
"Call-ID: 94bb2520-c0a80165-13c4-3e635-3ccd2971-3e635@rjolyscs2.ca.nortel.com\r\n"
"CSeq: 1 INVITE\r\n"
"Via: SIP/2.0/UDP 192.168.1.101:5060;received=47.135.162.145;rport=14956;branch=z9hG4bK-3e635-f3b41fc-310ddca7;id=1234-0\r\n"
"Supported: replaces\r\n"
"Contact: <sip:[email protected]:5060>\r\n"
"Record-Route: <sip:47.135.162.140:5060;lr;sipXecs-rs=%2Afrom%7EOTRiYzI1YjgtYzBhODAxNjUtMTNjNC0zZTYzNS0zN2FhMTk4OS0zZTYzNQ%60%60.400_authrules%2Aauth%7E.900_ntap%2ACeT%7ENDcuMTM1LjE2Mi4xNDU6Mjk1NDQ7dHJhbnNwb3J0PXVkcA%60%60.900_ntap%2ACrT%7ENDcuMTM1LjE2Mi4xNDU6MTQ5NTY7dHJhbnNwb3J0PXVkcA%60%60.900_ntap%2Aid%7EMC0w%215d4c53f8d9c9ecd8d089e57ba7a59513>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: 202\r\n"
"\r\n"
"\r\n"
"v=0\r\n"
"o=LGEIPP 5466 5466 IN IP4 192.168.1.11\r\n"
"s=SIP Call\r\n"
"c=IN IP4 192.168.1.11\r\n"
"t=0 0\r\n"
"m=audio 23020 RTP/AVP 0 101\r\n"
"a=rtpmap:0 PCMU/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
"a=fmtp:101 0-11\r\n"
"a=sendrecv\r\n"
"a=x-sipx-ntap:192.168.0.2;1\r\n";
SipMessage okResp( response, strlen(response) );
pNatTraversalAgent->handleOutputMessage( okResp, "47.135.162.145", 14956 );
// check that the record route still has our public IP address
CPPUNIT_ASSERT( okResp.getRecordRouteUri( 0, &tmpRecordRoute ) );
Url tmpRecordRouteUrl2( tmpRecordRoute );
tmpRecordRouteUrl2.getHostWithPort( tmpRecordRouteHost );
ASSERT_STR_EQUAL( "47.135.162.140:5060", tmpRecordRouteHost.data() );
// check that the SDP got changed properly to relay media
pSdpBody = okResp.getSdpBody();
CPPUNIT_ASSERT( pSdpBody );
CPPUNIT_ASSERT( pSdpBody->getMediaData( 0,
&mediaType,
&mediaPort,
&mediaPortPairs,
&mediaTransportType,
10,
&numPayloadTypes,
payloadTypes,
&dir ) == TRUE );
CPPUNIT_ASSERT( mediaType == "audio" );
CPPUNIT_ASSERT( mediaPort == 11000 );
CPPUNIT_ASSERT( mediaPortPairs == 1 );
CPPUNIT_ASSERT( mediaTransportType == "RTP/AVP" );
CPPUNIT_ASSERT( numPayloadTypes == 2 );
CPPUNIT_ASSERT( payloadTypes[0] == 0);
CPPUNIT_ASSERT( payloadTypes[1] == 101 );
CPPUNIT_ASSERT( dir == sdpDirectionalitySendRecv );
// check that we are using the public IP address of the media relay
CPPUNIT_ASSERT( pSdpBody->getMediaAddress(0, &mediaAddress ) );
ASSERT_STR_EQUAL( pNatTraversalAgent->mNatTraversalRules.getMediaRelayPublicAddress(), mediaAddress );
}
void handleLocalNATedToRemoteNATedCall()
{
const char* message =
"INVITE sip:[email protected]:5060;x-sipX-pubcontact=47.135.162.145%3A29544 SIP/2.0\r\n"
"Record-Route: <sip:192.168.0.2:5060;lr>\r\n"
"From: \"L3\"<sip:[email protected]>;tag=94bb9480-c0a80065-13c4-50045-20c55bd3-50045\r\n"
"To: <sip:[email protected]>\r\n"
"Call-Id: 94bb55a0-c0a80065-13c4-50045-61a3ad3-50045@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bK-sipXecs-8130adee90e0277761fec4ee1c194a56838f~9d6590763d78764b93a42e94dbd6b75a\r\n"
"Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-50045-13890e4e-4f17a229\r\n"
"Max-Forwards: 19\r\n"
"Supported: replaces\r\n"
"User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A183385\r\n"
"Contact: <sip:[email protected]:5060;x-sipX-nonat>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: 301\r\n"
"Date: Fri, 13 Jun 2008 14:48:44 GMT\r\n"
"\r\n"
"\r\n"
"v=0\r\n"
"o=LGEIPP 14595 14595 IN IP4 192.168.0.101\r\n"
"s=SIP Call\r\n"
"c=IN IP4 192.168.0.101\r\n"
"t=0 0\r\n"
"m=audio 23016 RTP/AVP 18 4 0 8 101\r\n"
"a=rtpmap:18 G729/8000\r\n"
"a=rtpmap:4 G723/8000\r\n"
"a=rtpmap:0 PCMU/8000\r\n"
"a=rtpmap:8 PCMA/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
"a=fmtp:18 annexb=no\r\n"
"a=fmtp:101 0-11\r\n"
"a=sendrecv\r\n";
SipMessage testMsg(message, strlen(message));
UtlString rejectReason;
UtlSList emptyList;
RouteState routeState( message, emptyList, "192.168.0.2:5060" );
AuthPlugin::AuthResult result;
result = pNatTraversalAgent->authorizeAndModify( "id", //dummy
Url("sip:[email protected]:5060"), //dummy
routeState,
"INVITE",
AuthPlugin::CONTINUE,
testMsg,
false,
rejectReason );
// check auth result
CPPUNIT_ASSERT( result == AuthPlugin::CONTINUE );
// check that call tracker got created
UtlString callId;
CallTracker* pTracker;
testMsg.getCallIdField( &callId );
pTracker = pNatTraversalAgent->getCallTrackerFromCallId( callId );
CPPUNIT_ASSERT( pTracker != 0 );
// check that no public transport of caller got encoded in route state
UtlString transportInfo;
CPPUNIT_ASSERT( !routeState.getParameter( pNatTraversalAgent->mInstanceName.data(), CALLER_PUBLIC_TRANSPORT_PARAM, transportInfo ) );
// check that public transport of callee got encoded in route state
routeState.getParameter( pNatTraversalAgent->mInstanceName.data(), CALLEE_PUBLIC_TRANSPORT_PARAM, transportInfo );
ASSERT_STR_EQUAL( "47.135.162.145:29544;transport=udp", transportInfo.data() );
// Check that sipX NAT route is set to reach the callee's public IP address
UtlString natRoute;
CPPUNIT_ASSERT( testMsg.getSipXNatRoute( &natRoute ) == true );
ASSERT_STR_EQUAL( "47.135.162.145:29544;transport=udp", natRoute.data() );
// Now present the message to the Output Processor and verify that it got processed correctly.
routeState.update( &testMsg ); // uodate route state as it would normally be done by SipRouter.
pNatTraversalAgent->handleOutputMessage( testMsg, "47.135.162.145", 29544 );
// verify that top via got adjusted to reflect our public IP address
UtlString topmostVia;
CPPUNIT_ASSERT( testMsg.getViaFieldSubField( &topmostVia, 0 ) );
ASSERT_STR_EQUAL( "SIP/2.0/UDP 47.135.162.140:5060;branch=z9hG4bK-sipXecs-8130adee90e0277761fec4ee1c194a56838f~9d6590763d78764b93a42e94dbd6b75a",
topmostVia.data() );
// verify that top route got modfified to reflect our public address
UtlString tmpRecordRoute;
CPPUNIT_ASSERT( testMsg.getRecordRouteUri( 0, &tmpRecordRoute ) );
Url tmpRecordRouteUrl( tmpRecordRoute );
UtlString tmpRecordRouteHost;
tmpRecordRouteUrl.getHostWithPort( tmpRecordRouteHost );
ASSERT_STR_EQUAL( "47.135.162.140:5060", tmpRecordRouteHost.data() );
// recall the call tracker based on message and compare with previous
CallTracker* pTracker2;
pTracker2 = pNatTraversalAgent->getCallTrackerForMessage( testMsg );
CPPUNIT_ASSERT( pTracker2 == pTracker );
// check that the SDP got changed properly to relay media
const SdpBody* pSdpBody = testMsg.getSdpBody();
CPPUNIT_ASSERT( pSdpBody );
UtlString mediaType, mediaTransportType;
int mediaPort, mediaPortPairs, numPayloadTypes, payloadTypes[10];
SdpDirectionality dir;
CPPUNIT_ASSERT( pSdpBody->getMediaData( 0,
&mediaType,
&mediaPort,
&mediaPortPairs,
&mediaTransportType,
10,
&numPayloadTypes,
payloadTypes,
&dir ) == TRUE );
CPPUNIT_ASSERT( mediaType == "audio" );
CPPUNIT_ASSERT( mediaPort == 10000 );
CPPUNIT_ASSERT( mediaPortPairs == 1 );
CPPUNIT_ASSERT( mediaTransportType == "RTP/AVP" );
CPPUNIT_ASSERT( numPayloadTypes == 5 );
CPPUNIT_ASSERT( payloadTypes[0] == 18 );
CPPUNIT_ASSERT( payloadTypes[1] == 4 );
CPPUNIT_ASSERT( payloadTypes[2] == 0 );
CPPUNIT_ASSERT( payloadTypes[3] == 8 );
CPPUNIT_ASSERT( payloadTypes[4] == 101 );
CPPUNIT_ASSERT( dir == sdpDirectionalitySendRecv );
// check that we are using the public IP address of the media relay
UtlString mediaAddress;
CPPUNIT_ASSERT( pSdpBody->getMediaAddress(0, &mediaAddress ) );
ASSERT_STR_EQUAL( pNatTraversalAgent->mNatTraversalRules.getMediaRelayPublicAddress(), mediaAddress );
// Mow simulalate the reception of the response
const char* response =
"SIP/2.0 200 OK\r\n"
"From: \"L3\"<sip:[email protected]>;tag=94bb9480-c0a80065-13c4-50045-20c55bd3-50045\r\n"
"To: <sip:[email protected]>;tag=94bf81c0-c0a8010b-13c4-4fc07-650a9211-4fc07\r\n"
"Call-Id: 94bb55a0-c0a80065-13c4-50045-61a3ad3-50045@rjolyscs2.ca.nortel.com\r\n"
"Cseq: 1 INVITE\r\n"
"Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-50045-13890e4e-4f17a229;id=1234-1\r\n"
"Supported: replaces\r\n"
"Contact: <sip:[email protected]:5060>\r\n"
"Record-Route: <sip:47.135.162.140:5060;lr;sipXecs-rs=%2Afrom%7EOTRiYjk0ODAtYzBhODAwNjUtMTNjNC01MDA0NS0yMGM1NWJkMy01MDA0NQ%60%60.400_authrules%2Aauth%7E.900_ntap%2ACeT%7ENDcuMTM1LjE2Mi4xNDU6Mjk1NDQ7dHJhbnNwb3J0PXVkcA%60%60.900_ntap%2Aid%7EMy0z%21641c6aa1807d3544000f310a64a5360a>\r\n"
"Content-Type: application/sdp\r\n"
"Content-Length: 250\r\n"
"Date: Fri, 13 Jun 2008 14:48:46 GMT\r\n"
"\r\n"
"\r\n"
"v=0\r\n"
"o=LGEIPP 12617 12617 IN IP4 192.168.1.11\r\n"
"s=SIP Call\r\n"
"c=IN IP4 192.168.1.11\r\n"
"t=0 0\r\n"
"m=audio 25038 RTP/AVP 4 101\r\n"
"c=IN IP4 192.168.0.2\r\n"
"a=rtpmap:4 G723/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
"a=fmtp:101 0-11\r\n"
"a=sendrecv\r\n"
"a=x-sipx-ntap:192.168.0.2;3\r\n";
SipMessage okResp( response, strlen(response) );
pNatTraversalAgent->handleOutputMessage( okResp, "192.168.0.101", 5060 );
// check that the record route has our private IP address
CPPUNIT_ASSERT( okResp.getRecordRouteUri( 0, &tmpRecordRoute ) );
Url tmpRecordRouteUrl2( tmpRecordRoute );
tmpRecordRouteUrl2.getHostWithPort( tmpRecordRouteHost );
ASSERT_STR_EQUAL( "192.168.0.2:5060", tmpRecordRouteHost.data() );
// check that the SDP got changed properly to relay media
pSdpBody = okResp.getSdpBody();
CPPUNIT_ASSERT( pSdpBody );
CPPUNIT_ASSERT( pSdpBody->getMediaData( 0,
&mediaType,
&mediaPort,
&mediaPortPairs,
&mediaTransportType,
10,
&numPayloadTypes,
payloadTypes,
&dir ) == TRUE );
CPPUNIT_ASSERT( mediaType == "audio" );
CPPUNIT_ASSERT( mediaPort == 11000 );
CPPUNIT_ASSERT( mediaPortPairs == 1 );
CPPUNIT_ASSERT( mediaTransportType == "RTP/AVP" );
CPPUNIT_ASSERT( numPayloadTypes == 2 );
CPPUNIT_ASSERT( payloadTypes[0] == 4);
CPPUNIT_ASSERT( payloadTypes[1] == 101 );
CPPUNIT_ASSERT( dir == sdpDirectionalitySendRecv );
// check that we are using the native IP address of the media relay
CPPUNIT_ASSERT( pSdpBody->getMediaAddress(0, &mediaAddress ) );
ASSERT_STR_EQUAL( pNatTraversalAgent->mNatTraversalRules.getMediaRelayNativeAddress(), mediaAddress );
}
void testLotsaRoutes()
{
const char* message =
"INVITE sip:example.com;lr SIP/2.0\r\n"
"Route: "
"<sip:internal.example.com;lr>, "
"<sip:external.example.net;lr>, "
"<sip:somewhere.else.net;lr>, "
"<sip:some.other.net;lr>, "
"<sip:somewhereelse.org;lr>, "
"<sip:some-other.net;lr>, "
"<sip:proxy.other.net;lr>, "
"<sip:[email protected]>\r\n"
"Via: SIP/2.0/TCP 10.1.1.3:33855\r\n"
"To: sip:[email protected]\r\n"
"From: Caller <sip:[email protected]>; tag=30543f3483e1cb11ecb40866edd3295b\r\n"
"Call-Id: f88dfabce84b6a2787ef024a7dbe8749\r\n"
"Cseq: 1 INVITE\r\n"
"Max-Forwards: 20\r\n"
"Contact: [email protected]\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg(message, strlen(message));
Url requestUri;
UtlSList removedRoutes;
testMsg.normalizeProxyRoutes(UserAgent, requestUri, &removedRoutes);
UtlString normalizedMsg;
int msgLen;
testMsg.getBytes(&normalizedMsg, &msgLen);
UtlString requestUriResult;
requestUri.toString(requestUriResult);
ASSERT_STR_EQUAL("sip:[email protected]", requestUriResult.data());
UtlString* removedRoute;
CPPUNIT_ASSERT( removedRoute = static_cast<UtlString*>(removedRoutes.get()));
ASSERT_STR_EQUAL("<sip:example.com;lr>", removedRoute->data());
delete removedRoute;
CPPUNIT_ASSERT( removedRoute = static_cast<UtlString*>(removedRoutes.get()));
ASSERT_STR_EQUAL("<sip:internal.example.com;lr>", removedRoute->data());
delete removedRoute;
CPPUNIT_ASSERT( removedRoute = static_cast<UtlString*>(removedRoutes.get()));
ASSERT_STR_EQUAL("<sip:external.example.net;lr>", removedRoute->data());
delete removedRoute;
CPPUNIT_ASSERT(removedRoutes.isEmpty());
UtlString route;
/* "Route: "
* "<sip:somewhere.else.net;lr>, "
* "<sip:some.other.net;lr>, "
* "<sip:somewhereelse.org;lr>, "
* "<sip:some-other.net;lr>, "
* "<sip:proxy.other.net;lr>"
*/
CPPUNIT_ASSERT(getNormalRoute(testMsg, 0, route));
ASSERT_STR_EQUAL("<sip:somewhere.else.net;lr>", route.data());
CPPUNIT_ASSERT(getNormalRoute(testMsg, 1, route));
ASSERT_STR_EQUAL("<sip:some.other.net;lr>", route.data());
CPPUNIT_ASSERT(getNormalRoute(testMsg, 2, route));
ASSERT_STR_EQUAL("<sip:somewhereelse.org;lr>", route.data());
CPPUNIT_ASSERT(getNormalRoute(testMsg, 3, route));
ASSERT_STR_EQUAL("<sip:some-other.net;lr>", route.data());
CPPUNIT_ASSERT(getNormalRoute(testMsg, 4, route));
ASSERT_STR_EQUAL("<sip:proxy.other.net;lr>", route.data());
CPPUNIT_ASSERT(!getNormalRoute(testMsg, 5, route));
}
void testShutdownNonBlocking()
{
int myPID = OsProcess::getCurrentPID();
int startingThreads;
// Stop TimerTask and NatAgentTask before counting threads.
// Some tests do not bother stopping them, so they may come started.
OsTimerTask::destroyTimerTask();
OsNatAgentTask::releaseInstance();
// Count number of threads now.
startingThreads = getNumThreads(myPID);
// Simple invite message from siptest/src/siptest/invite.txt
const char* SimpleMessage =
"INVITE sip:[email protected] SIP/2.0\r\n"
"Route: <sip:[email protected]:5064;lr>\r\n"
"From: <sip:[email protected];user=phone>;tag=bbb\r\n"
"To: <sip:[email protected]:3000;user=phone>\r\n"
"Call-Id: 8\r\n"
"Cseq: 1 INVITE\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg( SimpleMessage, strlen( SimpleMessage ) );
for(int i = 0; i < SHUTDOWN_TEST_ITERATIONS; ++i)
{
// Limit life time of lineMgr and refreshMgr. They should be freed
// before releasing OsNatAgentTask instance, or we will crash.
{
SipLineMgr lineMgr;
SipRefreshMgr refreshMgr;
lineMgr.StartLineMgr();
lineMgr.initializeRefreshMgr( &refreshMgr );
SipUserAgent sipUA( 5090
,5090
,5091
,NULL // default publicAddress
,NULL // default defaultUser
,"127.0.0.1" // default defaultSipAddress
,NULL // default sipProxyServers
,NULL // default sipDirectoryServers
,NULL // default sipRegistryServers
,NULL // default authenticationScheme
,NULL // default authenicateRealm
,NULL // default authenticateDb
,NULL // default authorizeUserIds
,NULL // default authorizePasswords
,&lineMgr
);
sipUA.start();
refreshMgr.init(&sipUA);
sipUA.send(testMsg);
// Wait long enough for some stack timeouts/retransmits to occur
OsTask::delay(10000); // 10 seconds
sipUA.shutdown(FALSE);
lineMgr.requestShutdown();
refreshMgr.requestShutdown();
while(!sipUA.isShutdownDone())
{
;
}
CPPUNIT_ASSERT(sipUA.isShutdownDone());
}
// Stop TimerTask and NatAgentTask again before counting threads.
// They were started while testing.
OsTimerTask::destroyTimerTask();
OsNatAgentTask::releaseInstance();
// Test to see that all the threads created by the above operations
// get properly shut down.
// Since the threads do not shut down synchronously with the above
// calls, we have to wait before we know they will be cleared.
OsTask::delay(1000); // 1 second
int numThreads = getNumThreads(myPID);
CPPUNIT_ASSERT_EQUAL(startingThreads,numThreads);
}
};
// Test that an out-of-dialog request gets a Record-Route, even if it does not
// require authorization/authentication, and test that the Record-Route has no
// extraneous parameters applied.
void testNoPermNeededOut()
{
OsConfigDb configuration;
configuration.set("RULES", TEST_DATA_DIR "/enforcerules.xml");
enforcer->readConfig(configuration);
UtlString identity; // no authenticated identity
Url requestUri("sip:911@emergency-gw");
const char* message =
"INVITE sip:911@emergency-gw SIP/2.0\r\n"
"Via: SIP/2.0/TCP 10.1.1.3:33855\r\n"
"To: sip:911@emergency-gw\r\n"
"From: Caller <sip:[email protected]>; tag=30543f3483e1cb11ecb40866edd3295b\r\n"
"Call-Id: f88dfabce84b6a2787ef024a7dbe8749\r\n"
"Cseq: 2 INVITE\r\n"
"Max-Forwards: 20\r\n"
"Contact: [email protected]\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg(message, strlen(message));
UtlSList noRemovedRoutes;
UtlString routeName("example.com");
RouteState routeState( testMsg, noRemovedRoutes, routeName );
const char unmodifiedRejectReason[] = "unmodified";
UtlString rejectReason(unmodifiedRejectReason);
UtlString method("INVITE");
const bool bSpiralingRequest = false;
AuthPlugin::AuthResult priorResult = AuthPlugin::CONTINUE;
CPPUNIT_ASSERT(AuthPlugin::ALLOW
== enforcer->authorizeAndModify(identity,
requestUri,
routeState,
method,
priorResult,
testMsg,
bSpiralingRequest,
rejectReason
));
ASSERT_STR_EQUAL(unmodifiedRejectReason, rejectReason.data());
// No Record-Route header.
routeState.update(&testMsg);
UtlString recordRoute;
CPPUNIT_ASSERT(!testMsg.getRecordRouteField(0, &recordRoute));
RouteState spiraledRouteState(testMsg, noRemovedRoutes, routeName);
// now simulate a spiral with the same message
CPPUNIT_ASSERT(AuthPlugin::ALLOW
== enforcer->authorizeAndModify(identity,
requestUri,
spiraledRouteState,
method,
priorResult,
testMsg,
bSpiralingRequest,
rejectReason
));
ASSERT_STR_EQUAL(unmodifiedRejectReason, rejectReason.data());
// No Record-Route header.
spiraledRouteState.update(&testMsg);
CPPUNIT_ASSERT(!testMsg.getRecordRouteField(0, &recordRoute));
}
// Test that an ACK is not challenged and not RecordRouted
void testNoPermAck()
{
OsConfigDb configuration;
configuration.set("RULES", TEST_DATA_DIR "/enforcerules.xml");
enforcer->readConfig(configuration);
UtlString identity; // no authenticated identity
Url requestUri("sip:somewhere@forbidden");
const char* message =
"ACK sip:somewhere@forbidden SIP/2.0\r\n"
"Via: SIP/2.0/TCP 10.1.1.3:33855\r\n"
"To: sip:somewhere@forbidden\r\n"
"From: Caller <sip:[email protected]>; tag=99911983748\r\n"
"Call-Id: b1373e736d7d359ead76fa5cd467d999\r\n"
"Cseq: 2 ACK\r\n"
"Max-Forwards: 20\r\n"
"Contact: [email protected]\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg(message, strlen(message));
UtlSList noRemovedRoutes;
UtlString routeName("example.com");
RouteState routeState( testMsg, noRemovedRoutes, routeName );
const char unmodifiedRejectReason[] = "unmodified";
UtlString rejectReason(unmodifiedRejectReason);
UtlString method("ACK");
const bool bSpiralingRequest = false;
AuthPlugin::AuthResult priorResult = AuthPlugin::ALLOW;
CPPUNIT_ASSERT(AuthPlugin::CONTINUE
== enforcer->authorizeAndModify(identity,
requestUri,
routeState,
method,
priorResult,
testMsg,
bSpiralingRequest,
rejectReason
));
ASSERT_STR_EQUAL(unmodifiedRejectReason, rejectReason.data());
routeState.update(&testMsg);
UtlString recordRoute;
CPPUNIT_ASSERT(!testMsg.getRecordRouteField(0, &recordRoute));
// now simulate a spiral with the same message
CPPUNIT_ASSERT(AuthPlugin::CONTINUE
== enforcer->authorizeAndModify(identity,
requestUri,
routeState,
method,
priorResult,
testMsg,
bSpiralingRequest,
rejectReason
));
ASSERT_STR_EQUAL(unmodifiedRejectReason, rejectReason.data());
routeState.update(&testMsg);
CPPUNIT_ASSERT(!testMsg.getRecordRouteField(0, &recordRoute));
}
void testShutdownBlocking()
{
pid_t myPID = OsProcess::getCurrentPID();
int startingThreads = getNumThreads(myPID);
// Simple invite message from siptest/src/siptest/invite.txt
const char* SimpleMessage =
"INVITE sip:[email protected] SIP/2.0\r\n"
"Route: <sip:[email protected]:5064;lr>\r\n"
"From: <sip:[email protected];user=phone>;tag=bbb\r\n"
"To: <sip:[email protected]:3000;user=phone>\r\n"
"Call-Id: 8\r\n"
"Cseq: 1 INVITE\r\n"
"Content-Length: 0\r\n"
"\r\n";
SipMessage testMsg( SimpleMessage, strlen( SimpleMessage ) );
for(int i = 0; i < SHUTDOWN_TEST_ITERATIONS; ++i)
{
{
SipLineMgr lineMgr;
SipRefreshMgr refreshMgr;
lineMgr.StartLineMgr();
lineMgr.initializeRefreshMgr( &refreshMgr );
SipUserAgent sipUA( 5090
,5090
,5091
,NULL // default publicAddress
,NULL // default defaultUser
,"127.0.0.1" // default defaultSipAddress
,NULL // default sipProxyServers
,NULL // default sipDirectoryServers
,NULL // default sipRegistryServers
,NULL // default authenticationScheme
,NULL // default authenicateRealm
,NULL // default authenticateDb
,NULL // default authorizeUserIds
,NULL // default authorizePasswords
,&lineMgr
);
sipUA.start();
refreshMgr.init(&sipUA);
sipUA.send(testMsg);
// Wait long enough for some stack timeouts/retansmits to occur
OsTask::delay(10000); // 10 seconds
// Shut down the tasks in reverse order.
refreshMgr.requestShutdown();
sipUA.shutdown(TRUE);
lineMgr.requestShutdown();
CPPUNIT_ASSERT(sipUA.isShutdownDone());
OsTimerTask::destroyTimerTask();
OsStunAgentTask::releaseInstance();
}
// Test to see that all the threads created by the above operations
// get properly shut down.
int numThreads = getNumThreads(myPID);
OsSysLog::add(FAC_SIP, PRI_NOTICE, "SipUserAgentTest::testShutdownBlocking "
"numThreads=%d startingThreads=%d",
numThreads, startingThreads);
KNOWN_BUG("XECS-48", "Some threads are not cleaned up?");
CPPUNIT_ASSERT(numThreads <= startingThreads);
}
};
