Added support for capturing audio from any Sangoma telephony board (Sangoma driver needs to be capable of generating RTP copy of voice traffic).

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@451 09dcff7a-b715-0410-9601-b79a96267cd0

4 files changed, 81 insertions, 8 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index 397874f..a51e00d 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -135,7 +135,14 @@ void RtpSession::ProcessMetadataRawRtp(RtpPacketInfoRef& rtpPacket)
 	char szDestIp[16];
 	ACE_OS::inet_ntop(AF_INET, (void*)&rtpPacket->m_destIp, szDestIp, sizeof(szDestIp));
 
-	m_capturePort = m_ipAndPort;
+	if(DLLCONFIG.m_sangomaEnable)
+	{
+		m_capturePort = IntToString(rtpPacket->m_sourcePort % 1000);
+	}
+	else
+	{
+		m_capturePort = m_ipAndPort;
+	}
 
 	if(sourceIsLocal)
 	{
@@ -621,7 +628,7 @@ bool RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 	if(m_log->isDebugEnabled())
 	{
 		CStdString debug;
-		debug.Format("[%s] %s: Add RTP packet ts:%u  arrival:%u ch:%d", m_trackingId, m_capturePort, rtpPacket->m_timestamp, rtpPacket->m_arrivalTimestamp, channel);
+		debug.Format("[%s] %s: Add RTP packet srcPort:%u dstPort:%u seq:%u ts:%u  arrival:%u ch:%d", m_trackingId, m_capturePort, rtpPacket->m_sourcePort, rtpPacket->m_destPort, rtpPacket->m_seqNum, rtpPacket->m_timestamp, rtpPacket->m_arrivalTimestamp, channel);
 		LOG4CXX_DEBUG(m_log, debug);
 	}
 
@@ -1267,9 +1274,27 @@ void RtpSessions::ReportRtpPacket(RtpPacketInfoRef& rtpPacket)
 	RtpSessionRef session;
 	CStdString logMsg;
 
-	if(rtpPacket->m_sourcePort == 1075)
+	int sourcePort = rtpPacket->m_sourcePort;
+	int destPort = rtpPacket->m_destPort;
+
+	if(DLLCONFIG.m_sangomaEnable && sourcePort == destPort)
 	{
-		int i=0;
+		if(sourcePort > DLLCONFIG.m_sangomaTxTcpPortStart)
+		{
+			// This is a TX packet
+			sourcePort = sourcePort - DLLCONFIG.m_sangomaTcpPortDelta;
+			rtpPacket->m_sourcePort = sourcePort;
+			// flip the least significant bit of the most significant byte
+			rtpPacket->m_sourceIp.s_addr ^= 0x00000001;
+		}
+		else
+		{
+			// This is an RX packet
+			sourcePort = sourcePort + DLLCONFIG.m_sangomaTcpPortDelta;
+			rtpPacket->m_sourcePort = sourcePort;
+			// flip the least significant bit of the most significant byte
+			rtpPacket->m_destIp.s_addr ^= 0x00000001;
+		}
 	}
 
 	// Add RTP packet to session with matching source or dest IP+Port. 
@@ -1277,7 +1302,7 @@ void RtpSessions::ReportRtpPacket(RtpPacketInfoRef& rtpPacket)
 	// phone call, so this RTP packet can potentially be reported to two sessions.
 
 	// Does a session exist with this source Ip+Port
-	CStdString port = IntToString(rtpPacket->m_sourcePort);
+	CStdString port = IntToString(sourcePort);
 	char szSourceIp[16];
 	ACE_OS::inet_ntop(AF_INET, (void*)&rtpPacket->m_sourceIp, szSourceIp, sizeof(szSourceIp));
 	CStdString sourceIpAndPort = CStdString(szSourceIp) + "," + port;
@@ -1304,7 +1329,7 @@ void RtpSessions::ReportRtpPacket(RtpPacketInfoRef& rtpPacket)
 	}
 
 	// Does a session exist with this destination Ip+Port
-	port = IntToString(rtpPacket->m_destPort);
+	port = IntToString(destPort);
 	char szDestIp[16];
 	ACE_OS::inet_ntop(AF_INET, (void*)&rtpPacket->m_destIp, szDestIp, sizeof(szDestIp));
 	CStdString destIpAndPort = CStdString(szDestIp) + "," + port;
@@ -1329,7 +1354,7 @@ void RtpSessions::ReportRtpPacket(RtpPacketInfoRef& rtpPacket)
 		}
 	}
 
-	if(numSessionsFound == 2)
+	if(numSessionsFound == 2 && session1.get() != session2.get())
 	{
 		// Need to "merge" the two sessions (ie discard one of them)
 		// Can happen when RTP stream detected before skinny StartMediaTransmission
@@ -1433,6 +1458,10 @@ void RtpSessions::ReportRtpPacket(RtpPacketInfoRef& rtpPacket)
 		{
 			ipAndPort = sourceIpAndPort;
 		}
+		else if(DLLCONFIG.m_sangomaEnable)
+		{
+			ipAndPort = sourceIpAndPort;
+		}
 		else
 		{
 			ipAndPort = destIpAndPort;
diff --git a/orkaudio/audiocaptureplugins/voip/VoIp.cpp b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
index 17c8d9e..c41d477 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
@@ -1022,7 +1022,7 @@ bool TryRtp(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, UdpH
 					rtpInfo->ToString(logMsg);
 					LOG4CXX_DEBUG(s_rtpPacketLog, logMsg);
 				}
-				if(payloadLength < 800)		// sanity check, speech RTP payload should always be smaller
+				if(payloadLength < 900)		// sanity check, speech RTP payload should always be smaller
 				{
 					RtpSessionsSingleton::instance()->ReportRtpPacket(rtpInfo);
 				}
diff --git a/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp b/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
index a079cd6..80aff5d 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
@@ -45,6 +45,10 @@ VoIpConfig::VoIpConfig()
 	m_skinnyIgnoreOpenReceiveChannelAck = false;
 	m_skinnyDynamicMediaAddress = false;
 	m_skinnyAllowCallInfoUpdate = false;
+
+	m_sangomaEnable = false;
+	m_sangomaRxTcpPortStart = 0;
+	m_sangomaTxTcpPortStart = 0;
 }
 
 void VoIpConfig::Define(Serializer* s)
@@ -82,6 +86,9 @@ void VoIpConfig::Define(Serializer* s)
 	s->BoolValue("SkinnyIgnoreOpenReceiveChannelAck", m_skinnyIgnoreOpenReceiveChannelAck);
 	s->BoolValue("SkinnyDynamicMediaAddress", m_skinnyDynamicMediaAddress);
 	s->BoolValue("SkinnyAllowCallInfoUpdate", m_skinnyAllowCallInfoUpdate);
+
+	s->IntValue("SangomaRxTcpPortStart", m_sangomaRxTcpPortStart);
+	s->IntValue("SangomaTxTcpPortStart", m_sangomaTxTcpPortStart);
 }
 
 void VoIpConfig::Validate()
@@ -249,6 +256,38 @@ void VoIpConfig::Validate()
                 exception.Format("VoIpConfig: RtpSessionOnHoldTimeOutSec must be > 0 (currently:%d) please fix config.xml", m_rtpSessionWithSignallingTimeoutSec);
                 throw (exception);
 	}
+
+	if(m_sangomaRxTcpPortStart == 0)
+	{
+	}
+	else if(m_sangomaRxTcpPortStart > 65000 || m_sangomaRxTcpPortStart < 2000 || ((m_sangomaRxTcpPortStart%1000) != 0) )
+	{
+		CStdString exception;
+		exception.Format("VoIpConfig: SangomaRxTcpPort must be between 2000 and 65000 and be a multiple of 1000 (currently:%d) please fix config.xml", m_sangomaRxTcpPortStart);
+		throw (exception);
+	}
+	if(m_sangomaTxTcpPortStart == 0)
+	{
+	}
+	else if(m_sangomaTxTcpPortStart > 65000 || m_sangomaTxTcpPortStart < 2000 || ((m_sangomaTxTcpPortStart%1000) != 0) )
+	{
+		CStdString exception;
+		exception.Format("VoIpConfig: SangomaTxTcpPort must be between 2000 and 65000 and be a multiple of 1000 (currently:%d) please fix config.xml", m_sangomaTxTcpPortStart);
+		throw (exception);
+	}
+
+	if(m_sangomaRxTcpPortStart > m_sangomaTxTcpPortStart)
+	{
+		CStdString exception;
+		exception.Format("VoIpConfig: SangomaTxTcpPort should always be bigger than SangomaRxTcpPort please fix config.xml");
+		throw (exception);
+	}
+	else if(m_sangomaRxTcpPortStart > 0 && m_sangomaTxTcpPortStart>0)
+	{
+		m_sangomaTcpPortDelta = m_sangomaTxTcpPortStart - m_sangomaRxTcpPortStart;
+		m_sangomaEnable = true;
+		m_rtpDetectOnOddPorts = true;
+	}
 }
 
 bool VoIpConfig::IsPartOfLan(struct in_addr addr)
diff --git a/orkaudio/audiocaptureplugins/voip/VoIpConfig.h b/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
index a9a4749..ea6186f 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
+++ b/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
@@ -81,6 +81,11 @@ public:
 
 	std::list<CStdString> m_dnisNumbers;
 	std::list<CStdString> m_sipExtractFields;
+
+	bool m_sangomaEnable;	// not a config parm, derived from the two following
+	int m_sangomaTcpPortDelta;
+	int m_sangomaRxTcpPortStart;
+	int m_sangomaTxTcpPortStart;
 };
 
 //========================================