Completed skinny support

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@25 09dcff7a-b715-0410-9601-b79a96267cd0

4 files changed, 166 insertions, 18 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h b/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h
index ae488bb..f108974 100644
--- a/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h
+++ b/orkaudio/audiocaptureplugins/voip/PacketHeaderDefs.h
@@ -135,6 +135,9 @@ typedef struct
 #define SKINNY_MSG_STOP_MEDIA_TRANSMISSION "StopMediaTransmission"
 #define SKINNY_MSG_CALL_INFO_MESSAGE "CallInfoMessage"
 
+#define SKINNY_CALL_TYPE_INBOUND 1
+#define SKINNY_CALL_TYPE_OUTBOUND 2
+
 typedef enum
 {
 	SkStartMediaTransmission = 0x008A,
diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index c8b49dd..c54c25b 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -44,11 +44,14 @@ RtpSession::RtpSession()
 
 void RtpSession::Stop()
 {
-	LOG4CXX_DEBUG(m_log, m_capturePort + " Session stop");
-	CaptureEventRef stopEvent(new CaptureEvent);
-	stopEvent->m_type = CaptureEvent::EtStop;
-	stopEvent->m_timestamp = time(NULL);
-	g_captureEventCallBack(stopEvent, m_capturePort);
+	if(m_started)
+	{
+		LOG4CXX_DEBUG(m_log, m_capturePort + " Session stop");
+		CaptureEventRef stopEvent(new CaptureEvent);
+		stopEvent->m_type = CaptureEvent::EtStop;
+		stopEvent->m_timestamp = time(NULL);
+		g_captureEventCallBack(stopEvent, m_capturePort);
+	}
 }
 
 void RtpSession::Start()
@@ -182,6 +185,18 @@ void RtpSession::ProcessMetadataSip(RtpPacketInfoRef& rtpPacket)
 	}
 }
 
+void RtpSession::ProcessMetadataSkinny(RtpPacketInfoRef& rtpPacket)
+{
+	// In skinny, we know that ipAndPort are those from the CallManager.
+	// However, what we want as a capture port are IP+Port of the phone
+	m_capturePort.Format("%s,%u", ACE_OS::inet_ntoa(rtpPacket->m_sourceIp), rtpPacket->m_sourcePort);
+	if(m_capturePort.Equals(m_ipAndPort))
+	{
+		m_capturePort.Format("%s,%u", ACE_OS::inet_ntoa(rtpPacket->m_destIp), rtpPacket->m_destPort);
+	}
+}
+
+
 void RtpSession::ReportMetadata()
 {
 	// report Local party
@@ -219,6 +234,12 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 			Start();
 			ReportMetadata();
 		}
+		else if(m_protocol == ProtSkinny)
+		{
+			ProcessMetadataSkinny(rtpPacket);
+			Start();
+			ReportMetadata();
+		}
 	}
 	m_lastRtpPacket = rtpPacket;
 	m_numRtpPackets++;
@@ -260,6 +281,10 @@ int RtpSession::ProtocolToEnum(CStdString& protocol)
 	{
 		protocolEnum = ProtSip;
 	}
+	else if (protocol.CompareNoCase(PROT_SKINNY) == 0)
+	{
+		protocolEnum = ProtSkinny;
+	}
 	return protocolEnum;
 }
 
@@ -274,6 +299,9 @@ CStdString RtpSession::ProtocolToString(int protocolEnum)
 	case ProtSip:
 		protocolString = PROT_SIP;
 		break;
+	case ProtSkinny:
+		protocolString = PROT_SKINNY;
+		break;
 	default:
 		protocolString = PROT_UNKN;
 	}
@@ -293,13 +321,20 @@ void RtpSessions::ReportSipInvite(SipInviteInfoRef& invite)
 	std::map<CStdString, RtpSessionRef>::iterator pair;
 	
 	pair = m_byIpAndPort.find(ipAndPort);
-
 	if (pair != m_byIpAndPort.end())
 	{
 		// A session exists ont the same IP+port, stop old session
 		RtpSessionRef session = pair->second;
 		Stop(session);
 	}
+	pair = m_byCallId.find(invite->m_callId);
+	if (pair != m_byCallId.end())
+	{
+		// A session exists ont the same CallId, stop old session
+		RtpSessionRef session = pair->second;
+		Stop(session);
+	}
+
 	// create new session and insert into both maps
 	RtpSessionRef session(new RtpSession());
 	session->m_ipAndPort = ipAndPort;
@@ -323,6 +358,87 @@ void RtpSessions::ReportSipBye(SipByeInfo bye)
 	}
 }
 
+void RtpSessions::ReportSkinnyCallInfo(SkCallInfoStruct* callInfo)
+{
+	CStdString callId = IntToString(callInfo->callId);
+	std::map<CStdString, RtpSessionRef>::iterator pair;
+	pair = m_byCallId.find(callId);
+	if (pair != m_byCallId.end())
+	{
+		// A session exists ont the same CallId, stop old session
+		RtpSessionRef session = pair->second;
+		Stop(session);
+	}
+
+	// create new session and insert into both maps
+	RtpSessionRef session(new RtpSession());
+	session->m_callId = callId;
+	session->m_protocol = RtpSession::ProtSkinny;
+	switch(callInfo->callType)
+	{
+	case SKINNY_CALL_TYPE_INBOUND:
+		session->m_localParty = callInfo->calledParty;
+		session->m_remoteParty = callInfo->callingParty;
+		session->m_direction = CaptureEvent::DirIn;
+		break;
+	case SKINNY_CALL_TYPE_OUTBOUND:
+		session->m_localParty = callInfo->callingParty;
+		session->m_remoteParty = callInfo->calledParty;
+		session->m_direction = CaptureEvent::DirOut;
+		break;
+	}
+	m_byCallId.insert(std::make_pair(session->m_callId, session));
+}
+
+void RtpSessions::ReportSkinnyStartMediaTransmission(SkStartMediaTransmissionStruct* startMedia)
+{
+	// Lookup by callId
+	CStdString callId = IntToString(startMedia->conferenceId);
+	std::map<CStdString, RtpSessionRef>::iterator pair;
+	pair = m_byCallId.find(callId);
+
+	if (pair != m_byCallId.end())
+	{
+		// Session found
+		RtpSessionRef session = pair->second;
+
+		if(session->m_ipAndPort.size() == 0)
+		{
+			CStdString ipAndPort;
+			ipAndPort.Format("%s,%u", ACE_OS::inet_ntoa(startMedia->remoteIpAddr), startMedia->remoteTcpPort);
+			
+			pair = m_byIpAndPort.find(ipAndPort);
+			if (pair != m_byIpAndPort.end())
+			{
+				// A session exists ont the same IP+port, stop old session
+				RtpSessionRef session = pair->second;
+				Stop(session);
+			}
+			session->m_ipAndPort = ipAndPort;
+			m_byIpAndPort.insert(std::make_pair(session->m_ipAndPort, session));
+		}
+		else
+		{
+			// The session has already had a StartMediaTransmission message.
+		}
+	}
+}
+
+void RtpSessions::ReportSkinnyStopMediaTransmission(SkStopMediaTransmissionStruct* stopMedia)
+{
+	CStdString callId = IntToString(stopMedia->conferenceId);
+
+	std::map<CStdString, RtpSessionRef>::iterator pair;
+	pair = m_byCallId.find(callId);
+
+	if (pair != m_byCallId.end())
+	{
+		// Session found: stop it
+		RtpSessionRef session = pair->second;
+		Stop(session);
+	}
+}
+
 void RtpSessions::Stop(RtpSessionRef& session)
 {
 	session->Stop();
@@ -381,7 +497,7 @@ void RtpSessions::Hoover(time_t now)
 	CStdString numSessions = IntToString(m_byIpAndPort.size());
 	LOG4CXX_DEBUG(m_log, "Hoover - check " + numSessions + " sessions time:" + IntToString(now));
 
-	// Go round the sessions and find inactive ones
+	// Go round the ipAndPort session index and find inactive sessions
 	std::map<CStdString, RtpSessionRef>::iterator pair;
 	std::list<RtpSessionRef> toDismiss;
 
@@ -401,5 +517,24 @@ void RtpSessions::Hoover(time_t now)
 		LOG4CXX_DEBUG(m_log, session->m_ipAndPort + " Expired");
 		Stop(session);
 	}
+
+	// Go round the callId session index and find inactive sessions
+	toDismiss.clear();
+	for(pair = m_byCallId.begin(); pair != m_byCallId.end(); pair++)
+	{
+		RtpSessionRef session = pair->second;
+		if((now - session->m_lastUpdated) > 10)
+		{
+			toDismiss.push_back(session);
+		}
+	}
+
+	// discard inactive sessions
+	for (it = toDismiss.begin(); it != toDismiss.end() ; it++)
+	{
+		RtpSessionRef session = *it;
+		LOG4CXX_DEBUG(m_log, session->m_ipAndPort + " Expired");
+		Stop(session);
+	}
 }
 
diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.h b/orkaudio/audiocaptureplugins/voip/RtpSession.h
index 3fd07c1..c3caadc 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.h
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.h
@@ -18,6 +18,7 @@
 #include "Rtp.h"
 #include <map>
 #include "ace/Singleton.h"
+#include "PacketHeaderDefs.h"
 
 class SipInviteInfo
 {
@@ -67,8 +68,9 @@ class RtpSession
 public:
 #define PROT_RAW_RTP "RawRtp"
 #define PROT_SIP "Sip"
+#define PROT_SKINNY "Skinny"
 #define PROT_UNKN "Unkn"
-	typedef enum{ProtRawRtp, ProtSip, ProtUnkn} ProtocolEnum;
+	typedef enum{ProtRawRtp, ProtSip, ProtSkinny, ProtUnkn} ProtocolEnum;
 	static int ProtocolToEnum(CStdString& protocol);
 	static CStdString ProtocolToString(int protocolEnum);
 
@@ -78,16 +80,21 @@ public:
 	void AddRtpPacket(RtpPacketInfoRef& rtpPacket);
 	void ReportSipInvite(SipInviteInfoRef& invite);
 
-	CStdString m_ipAndPort;
+	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)
 	CStdString m_callId;
 	SipInviteInfoRef m_invite;
 	time_t m_lastUpdated;
 	ProtocolEnum m_protocol;
+	CStdString m_localParty;
+	CStdString m_remoteParty;
+	CaptureEvent::DirectionEnum m_direction;
+
 private:
 	void ProcessMetadataSip(RtpPacketInfoRef&);
 	void ProcessMetadataSipIncoming();
 	void ProcessMetadataSipOutgoing();
 	void ProcessMetadataRawRtp(RtpPacketInfoRef&);
+	void ProcessMetadataSkinny(RtpPacketInfoRef& rtpPacket);
 	void ReportMetadata();
 
 	RtpPacketInfoRef m_lastRtpPacket;
@@ -99,9 +106,6 @@ private:
 	int m_inviteeTcpPort;
 	LoggerPtr m_log;
 	CStdString m_capturePort;
-	CStdString m_localParty;
-	CStdString m_remoteParty;
-	CaptureEvent::DirectionEnum m_direction;
 	bool m_started;
 };
 typedef boost::shared_ptr<RtpSession> RtpSessionRef;
@@ -114,6 +118,9 @@ public:
 	void Stop(RtpSessionRef& session);
 	void ReportSipInvite(SipInviteInfoRef& invite);
 	void ReportSipBye(SipByeInfo bye);
+	void ReportSkinnyCallInfo(SkCallInfoStruct*);
+	void ReportSkinnyStartMediaTransmission(SkStartMediaTransmissionStruct*);
+	void ReportSkinnyStopMediaTransmission(SkStopMediaTransmissionStruct*);
 	void ReportRtpPacket(RtpPacketInfoRef& rtpPacket);
 	void Hoover(time_t now);
 private:
diff --git a/orkaudio/audiocaptureplugins/voip/VoIp.cpp b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
index b2f83bf..adc21da 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
@@ -268,25 +268,27 @@ void HandleSkinnyMessage(SkinnyHeaderStruct* skinnyHeader)
 {
 	bool useful = true;
 	CStdString debug;
-	SkStartMediaTransmissionStruct* start;
-	SkStopMediaTransmissionStruct* stop;
+	SkStartMediaTransmissionStruct* startMedia;
+	SkStopMediaTransmissionStruct* stopMedia;
 	SkCallInfoStruct* callInfo;
 
 	switch(skinnyHeader->messageType)
 	{
 	case SkStartMediaTransmission:
-		start = (SkStartMediaTransmissionStruct*)skinnyHeader;
+		startMedia = (SkStartMediaTransmissionStruct*)skinnyHeader;
 		if(s_skinnyLog->isDebugEnabled())
 		{
-			debug.Format(" CallId:%u %s,%u", start->conferenceId, ACE_OS::inet_ntoa(start->remoteIpAddr), start->remoteTcpPort);
+			debug.Format(" CallId:%u %s,%u", startMedia->conferenceId, ACE_OS::inet_ntoa(startMedia->remoteIpAddr), startMedia->remoteTcpPort);
 		}
+		RtpSessionsSingleton::instance()->ReportSkinnyStartMediaTransmission(startMedia);
 		break;
 	case SkStopMediaTransmission:
-		stop = (SkStopMediaTransmissionStruct*)skinnyHeader;
+		stopMedia = (SkStopMediaTransmissionStruct*)skinnyHeader;
 		if(s_skinnyLog->isDebugEnabled())
 		{
-			debug.Format(" CallId:%u", stop->conferenceId);
+			debug.Format(" CallId:%u", stopMedia->conferenceId);
 		}
+		RtpSessionsSingleton::instance()->ReportSkinnyStopMediaTransmission(stopMedia);
 		break;
 	case SkCallInfoMessage:
 		callInfo = (SkCallInfoStruct*)skinnyHeader;
@@ -294,6 +296,7 @@ void HandleSkinnyMessage(SkinnyHeaderStruct* skinnyHeader)
 		{
 			debug.Format(" CallId:%u calling:%s called:%s", callInfo->callId, callInfo->callingParty, callInfo->calledParty);
 		}
+		RtpSessionsSingleton::instance()->ReportSkinnyCallInfo(callInfo);
 		break;
 	default:
 		useful = false;