Added ToString method to RtpPacketInfo class + RTP session now logs first packets for both ways on INFO level.

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@106 09dcff7a-b715-0410-9601-b79a96267cd0

4 files changed, 32 insertions, 5 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/Rtp.cpp b/orkaudio/audiocaptureplugins/voip/Rtp.cpp
index 931aed7..23c236e 100644
--- a/orkaudio/audiocaptureplugins/voip/Rtp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/Rtp.cpp
@@ -27,6 +27,16 @@ extern "C"
 extern AudioChunkCallBackFunction g_audioChunkCallBack;
 
 
+void RtpPacketInfo::ToString(CStdString& string)
+{
+	char sourceIp[16];
+	ACE_OS::inet_ntop(AF_INET, (void*)&m_sourceIp, sourceIp, sizeof(sourceIp));
+	char destIp[16];
+	ACE_OS::inet_ntop(AF_INET, (void*)&m_destIp, destIp, sizeof(destIp));
+	string.Format("%s,%d %s,%d seq:%u ts:%u len:%d type:%x", sourceIp, m_sourcePort, destIp, m_destPort, m_seqNum, m_timestamp, m_payloadSize, m_payloadType);
+}
+
+
 RtpRingBuffer::RtpRingBuffer()
 {
 	m_writePtr = m_buffer;
diff --git a/orkaudio/audiocaptureplugins/voip/Rtp.h b/orkaudio/audiocaptureplugins/voip/Rtp.h
index e64fd7d..a846c5f 100644
--- a/orkaudio/audiocaptureplugins/voip/Rtp.h
+++ b/orkaudio/audiocaptureplugins/voip/Rtp.h
@@ -28,6 +28,8 @@
 class RtpPacketInfo
 {
 public:
+	void ToString(CStdString& string);
+
 	//CStdString m_sourceMac;
 	//CStdString m_destMac;
 	struct in_addr m_sourceIp;
diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index d47361f..5a7a455 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -235,6 +235,8 @@ void RtpSession::ReportMetadata()
 
 void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 {
+	CStdString logMsg;
+
 	if(m_lastRtpPacket.get() == NULL)
 	{
 		// Until now, we knew the remote IP and port for RTP (at best, as given by SIP invite or Skinny StartMediaTransmission)
@@ -257,7 +259,14 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 	m_numRtpPackets++;
 	if(m_lastRtpPacketSide1.get() == NULL)
 	{
+		// First RTP packet for side 1
 		m_lastRtpPacketSide1 = rtpPacket;
+		if(m_log->isInfoEnabled())
+		{
+			rtpPacket->ToString(logMsg);
+			logMsg = "1st packet s1: " + logMsg;
+			LOG4CXX_INFO(m_log, logMsg);
+		}
 	}
 	else
 	{
@@ -267,6 +276,16 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 		}
 		else
 		{
+			if(m_lastRtpPacketSide2.get() == NULL)
+			{
+				// First RTP packet for side 2
+				if(m_log->isInfoEnabled())
+				{
+					rtpPacket->ToString(logMsg);
+					logMsg = "1st packet s2: " + logMsg;
+					LOG4CXX_INFO(m_log, logMsg);
+				}
+			}
 			m_lastRtpPacketSide2 = rtpPacket;
 		}
 	}
diff --git a/orkaudio/audiocaptureplugins/voip/VoIp.cpp b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
index 30dcc47..ce2f883 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
@@ -169,11 +169,7 @@ bool TryRtp(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, UdpH
 				if(s_rtpPacketLog->isDebugEnabled())
 				{
 					CStdString debug;
-					char sourceIp[16];
-					ACE_OS::inet_ntop(AF_INET, (void*)&ipHeader->ip_src, sourceIp, sizeof(sourceIp));
-					char destIp[16];
-					ACE_OS::inet_ntop(AF_INET, (void*)&ipHeader->ip_dest, destIp, sizeof(destIp));
-					debug.Format("%s,%d %s,%d seq:%u ts:%u len:%d", sourceIp, rtpInfo->m_sourcePort, destIp, rtpInfo->m_destPort, ntohs(rtpHeader->seq), ntohl(rtpHeader->ts), payloadLength);
+					rtpInfo->ToString(debug);
 					LOG4CXX_DEBUG(s_rtpPacketLog, debug);
 				}