1. Subsequent SIP INVITES that could be associated to an existing session now disregarded because they could be disrupting valid sessions. We need to store the new INVITES and only use them when they are validated by a matching RTP stream.

2. New RTP streams are now logged within one session by the VoIP plugin.
3. Added a Debug config boolean that has the trackingId alpha counter reset to 0 (AAA) when enabled.

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@380 09dcff7a-b715-0410-9601-b79a96267cd0

1 files changed, 2 insertions, 1 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.h b/orkaudio/audiocaptureplugins/voip/RtpSession.h
index d7f9659..77db42d 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.h
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.h
@@ -117,6 +117,7 @@ private:
 	unsigned int m_highestRtpSeqNumDelta;
 	double m_minRtpSeqDelta;
 	double m_minRtpTimestampDelta;
+	TcpAddressList m_rtpAddressList;
 };
 typedef boost::shared_ptr<RtpSession> RtpSessionRef;
 
@@ -147,7 +148,7 @@ private:
 	std::map<CStdString, RtpSessionRef> m_byCallId;
 	std::map<unsigned int, EndpointInfoRef> m_endpoints;
 	LoggerPtr m_log;
-	AlphaCounter alphaCounter;
+	AlphaCounter m_alphaCounter;
 };
 typedef ACE_Singleton<RtpSessions, ACE_Thread_Mutex> RtpSessionsSingleton;