Added support for SIP 200 OK message with ability to adjust the session RTP IP address and port if NAT is detected

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@474 09dcff7a-b715-0410-9601-b79a96267cd0

4 files changed, 116 insertions, 4 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h b/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h
index f054c43..ad6906d 100644
--- a/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h
+++ b/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h
@@ -337,8 +337,10 @@ struct Iax2MetaTrunkEntryTs {
 
 #define SIP_METHOD_INVITE_SIZE 6
 #define SIP_METHOD_BYE_SIZE 3
+#define SIP_RESPONSE_200_OK_SIZE 11
 #define SIP_METHOD_INVITE "INVITE"
 #define SIP_METHOD_BYE "BYE"
+#define SIP_RESPONSE_200_OK "SIP/2.0 200"
 
 #endif
 
diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index 91a483e..e29ad76 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -697,6 +697,7 @@ void RtpSession::ReportSipInvite(SipInviteInfoRef& invite)
 		invite->ToString(inviteString);
 		CStdString logMsg;
 		logMsg.Format("[%s] associating INVITE:%s", m_trackingId, inviteString);
+		m_fromRtpIp = invite->m_fromRtpIp;
 		LOG4CXX_INFO(m_log, logMsg);
 	}
 	m_invites.push_front(invite);
@@ -916,6 +917,41 @@ void RtpSessions::ReportSipErrorPacket(SipFailureMessageInfoRef& info)
 	return;
 }
 
+void RtpSessions::ReportSip200Ok(Sip200OkInfoRef info)
+{
+	std::map<CStdString, RtpSessionRef>::iterator pair;
+
+	pair = m_byCallId.find(info->m_callId);
+	if (pair != m_byCallId.end())
+	{
+		RtpSessionRef session = pair->second;
+
+		if(info->m_hasSdp && DLLCONFIG.IsPartOfLan(session->m_fromRtpIp) && !DLLCONFIG.IsPartOfLan(info->m_fromRtpIp) && !session->m_numRtpPackets)
+		{
+			char szFromRtpIp[16];
+
+			ACE_OS::inet_ntop(AF_INET, (void*)&info->m_fromRtpIp, szFromRtpIp, sizeof(szFromRtpIp));
+			m_byIpAndPort.erase(session->m_ipAndPort);
+			session->m_ipAndPort = CStdString(szFromRtpIp) + "," + info->m_fromRtpPort;
+
+			pair = m_byIpAndPort.find(session->m_ipAndPort);
+			if (pair != m_byIpAndPort.end())
+			{
+				RtpSessionRef session2 = pair->second;
+
+				if(session2->m_protocol == RtpSession::ProtRawRtp)
+				{
+					Stop(session2);
+				}
+			}
+
+			m_byIpAndPort.insert(std::make_pair(session->m_ipAndPort, session));
+
+			LOG4CXX_INFO(m_log, "[" + session->m_trackingId + "] updated RTP address: " + session->m_ipAndPort);
+		}
+	}
+}
+
 void RtpSessions::ReportSipBye(SipByeInfo bye)
 {
 	std::map<CStdString, RtpSessionRef>::iterator pair;
@@ -1751,3 +1787,9 @@ void SipFailureMessageInfo::ToString(CStdString& string)
 	string.Format("sender:%s rcvr:%s smac:%s rmac:%s callid:%s errorcode:%s errorstr:\"%s\"", senderIp, receiverIp, senderMac, receiverMac, m_callId, m_errorCode, m_errorString);
 }
 
+Sip200OkInfo::Sip200OkInfo()
+{
+	m_fromRtpIp.s_addr = 0;
+	m_hasSdp = false;
+}
+
diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.h b/orkaudio/audiocaptureplugins/voip/RtpSession.h
index 5475cb9..ec9d262 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.h
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.h
@@ -70,6 +70,20 @@ public:
 	CStdString m_callId;
 };
 
+
+class Sip200OkInfo
+{
+public:
+	Sip200OkInfo();
+	CStdString m_callId;
+	bool m_hasSdp;
+	struct in_addr m_fromRtpIp;
+	CStdString m_fromRtpPort;
+};
+
+typedef boost::shared_ptr<Sip200OkInfo> Sip200OkInfoRef;
+
+
 //=============================================================
 
 class EndpointInfo
@@ -104,6 +118,7 @@ public:
 
 	CStdString m_trackingId;
 	CStdString m_ipAndPort;	// IP address and TCP port of one side of the session, serves as a key for session storage and retrieval. Not necessarily the same as the capturePort (capturePort is usually the client (phone) IP+port)
+	struct in_addr m_fromRtpIp;
 	CStdString m_callId;
 	SipInviteInfoRef m_invite;
 	ACE_Time_Value m_creationDate;		// When the session is first created
@@ -180,6 +195,7 @@ public:
 	void ReportSkinnySoftKeyResume(SkSoftKeyEventMessageStruct* skEvent, IpHeaderStruct* ipHeader);
 	void ReportRtpPacket(RtpPacketInfoRef& rtpPacket);
 	void ReportSipErrorPacket(SipFailureMessageInfoRef& sipError);
+	void ReportSip200Ok(Sip200OkInfoRef info);
 	void Hoover(time_t now);
 	EndpointInfoRef GetEndpointInfo(struct in_addr endpointIp);
 	void StartCapture(CStdString& party);
diff --git a/orkaudio/audiocaptureplugins/voip/VoIp.cpp b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
index 2db26d7..60bcf0d 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
@@ -1360,6 +1360,55 @@ bool TrySipTcp(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, T
 	return result;
 }
 
+bool TrySip200Ok(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, UdpHeaderStruct* udpHeader, u_char* udpPayload, u_char* packetEnd)
+{
+	bool result = false;
+
+	int sipLength = ntohs(udpHeader->len) - sizeof(UdpHeaderStruct);
+	char* sipEnd = (char*)udpPayload + sipLength;
+	if(sipLength < SIP_RESPONSE_200_OK_SIZE || sipEnd > (char*)packetEnd)
+	{
+		;	// packet too short
+	}
+	else if (memcmp(SIP_RESPONSE_200_OK, (void*)udpPayload, SIP_RESPONSE_200_OK_SIZE) == 0)
+	{
+		result = true;
+
+		Sip200OkInfoRef info(new Sip200OkInfo());
+
+		char* callIdField = memFindAfter("\r\ni:", (char*)udpPayload, sipEnd);
+		char* audioField = NULL;
+		char* connectionAddressField = NULL;
+
+		if(callIdField)
+		{
+			GrabTokenSkipLeadingWhitespaces(callIdField, sipEnd, info->m_callId);
+			audioField = memFindAfter("m=audio ", callIdField, sipEnd);
+			connectionAddressField = memFindAfter("c=IN IP4 ", callIdField, sipEnd);
+		}
+		if(audioField && connectionAddressField)
+		{
+			info->m_hasSdp = true;
+
+			GrabToken(audioField, sipEnd, info->m_fromRtpPort);
+
+			CStdString connectionAddress;
+			GrabToken(connectionAddressField, sipEnd, connectionAddress);
+			struct in_addr fromIp;
+			if(connectionAddress.size())
+			{
+				if(ACE_OS::inet_aton((PCSTR)connectionAddress, &fromIp))
+				{
+					info->m_fromRtpIp = fromIp;
+				}
+			}
+		}
+
+		RtpSessionsSingleton::instance()->ReportSip200Ok(info);
+	}
+	return result;
+}
+
 bool TrySipInvite(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, UdpHeaderStruct* udpHeader, u_char* udpPayload, u_char* packetEnd)
 {
 	bool result = false;
@@ -1367,12 +1416,11 @@ bool TrySipInvite(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader
         
 	int sipLength = ntohs(udpHeader->len) - sizeof(UdpHeaderStruct);
 	char* sipEnd = (char*)udpPayload + sipLength;
-	if(sipLength < sizeof(SIP_METHOD_INVITE) || sipEnd > (char*)packetEnd)
+	if(sipLength < SIP_METHOD_INVITE_SIZE || sipEnd > (char*)packetEnd)
 	{
-		return false;
+		;	// packet too short
 	}
-
-	if (memcmp(SIP_METHOD_INVITE, (void*)udpPayload, SIP_METHOD_INVITE_SIZE) == 0)
+	else if (memcmp(SIP_METHOD_INVITE, (void*)udpPayload, SIP_METHOD_INVITE_SIZE) == 0)
 	{
 		result = true;
 
@@ -1828,6 +1876,10 @@ void HandlePacket(u_char *param, const struct pcap_pkthdr *header, const u_char
 			}
 
 			if(!detectedUsefulPacket) {
+				detectedUsefulPacket= TrySip200Ok(ethernetHeader, ipHeader, udpHeader, udpPayload, ipPacketEnd);
+			}
+
+			if(!detectedUsefulPacket) {
 				detectedUsefulPacket = TrySipBye(ethernetHeader, ipHeader, udpHeader, udpPayload, ipPacketEnd);
 			}