RTP sessions now check the destination IP address of a packet to determine whether it's side 1. This allows incoming PSTN calls transferred to various IP endpoints to be recorded as one session under two conditions: 1. There is no signalling for the second and subsequent segments and 2. the first RTP packet is sent from an internal endpoint to the PSTN gateway, see remark (1) in RtpSession.cpp.

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@370 09dcff7a-b715-0410-9601-b79a96267cd0

2 files changed, 5 insertions, 20 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index dc16115..7714d5d 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -43,7 +43,6 @@ RtpSession::RtpSession(CStdString& trackingId)
 	m_numRtpPackets = 0;
 	m_started = false;
 	m_stopped = false;
-	m_rtpTimestampCorrectiveDelta = 0;
 	m_beginDate = 0;
 	m_hasDuplicateRtp = false;
 	m_highestRtpSeqNumDelta = 0;
@@ -326,7 +325,6 @@ bool RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 {
 	CStdString logMsg;
 	unsigned char channel = 0;
-	unsigned int correctedRtpTimestamp = rtpPacket->m_timestamp;
 
 	// Dismiss packets that should not be part of a Skinny session
 	if(m_protocol == ProtSkinny)
@@ -372,7 +370,8 @@ bool RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 	}
 	else
 	{
-		if((unsigned int)rtpPacket->m_sourceIp.s_addr == (unsigned int)m_lastRtpPacketSide1->m_sourceIp.s_addr)
+		// Comparing destination IP address to find out if side1, see (1)
+		if((unsigned int)rtpPacket->m_destIp.s_addr == (unsigned int)m_lastRtpPacketSide1->m_destIp.s_addr)
 		{
 			if(rtpPacket->m_timestamp == m_lastRtpPacketSide1->m_timestamp)
 			{
@@ -454,22 +453,10 @@ bool RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 	if(m_log->isDebugEnabled())
 	{
 		CStdString debug;
-		debug.Format("[%s] %s: Add RTP packet ts:%u, corrected ts:%u, arrival:%u, channel:%d", m_trackingId, m_capturePort, rtpPacket->m_timestamp, correctedRtpTimestamp, rtpPacket->m_arrivalTimestamp, channel);
+		debug.Format("[%s] %s: Add RTP packet ts:%u  arrival:%u ch:%d", m_trackingId, m_capturePort, rtpPacket->m_timestamp, rtpPacket->m_arrivalTimestamp, channel);
 		LOG4CXX_DEBUG(m_log, debug);
 	}
 
-	// Detect RTP timestamp discontinuity
-	//if(m_protocol == ProtRawRtp)
-	//{
-	//	int delta = rtpPacket->m_timestamp - m_lastRtpPacketSide1->m_timestamp;
-	//	if(delta > (RTP_SESSION_TIMEOUT*8000) || delta < (RTP_SESSION_TIMEOUT*8000*-1))
-	//	{
-	//		logMsg.Format("RTP timestamp discontinuity before:%u after:%u", m_lastRtpPacketSide1->m_timestamp, rtpPacket->m_timestamp);
-	//		LOG4CXX_INFO(m_log, logMsg);
-	//		return false;
-	//	}
-	//}
-
 	if(		(m_protocol == ProtRawRtp && m_numRtpPackets == 50)	||
 			(m_protocol == ProtSkinny && m_numRtpPackets == 2)	||
 			(m_protocol == ProtSip && m_numRtpPackets == 2)			)
@@ -485,7 +472,7 @@ bool RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 		AudioChunkDetails details;
 		details.m_arrivalTimestamp = rtpPacket->m_arrivalTimestamp;
 		details.m_numBytes = rtpPacket->m_payloadSize;
-		details.m_timestamp = correctedRtpTimestamp;
+		details.m_timestamp = rtpPacket->m_timestamp;
 		details.m_rtpPayloadType = rtpPacket->m_payloadType;
 		details.m_sequenceNumber = rtpPacket->m_seqNum;
 		details.m_channel = channel;
@@ -1009,7 +996,7 @@ void RtpSessions::ReportRtpPacket(RtpPacketInfoRef& rtpPacket)
 		CStdString trackingId = alphaCounter.GetNext();
 		RtpSessionRef session(new RtpSession(trackingId));
 		session->m_protocol = RtpSession::ProtRawRtp;
-		session->m_ipAndPort = ipAndPort;
+		session->m_ipAndPort = ipAndPort;	// (1) In the case of a PSTN Gateway automated answer, This is the destination IP+Port of the first packet which is good, because it is usually the IP+Port of the PSTN Gateway.
 		session->AddRtpPacket(rtpPacket);
 		m_byIpAndPort.insert(std::make_pair(ipAndPort, session));
 
diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.h b/orkaudio/audiocaptureplugins/voip/RtpSession.h
index f1e0016..d7f9659 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.h
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.h
@@ -111,8 +111,6 @@ private:
 	CStdString m_capturePort;
 	bool m_started;
 	bool m_stopped;
-	unsigned int m_rtpTimestampCorrectiveDelta;
-	bool m_rtpTimestampCorrectiveSign;
 	time_t m_beginDate;
 	CStdString m_orkUid;
 	bool m_hasDuplicateRtp;