RTP session timeout and RTP session with signalling timeout (SIP, Skinny) are now configurable in config.xml: <VoIpPlugin><RtpSessionTimeoutSec> and <RtpSessionWithSignallingTimeoutSec></VoIpPlugin>

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@348 09dcff7a-b715-0410-9601-b79a96267cd0

3 files changed, 23 insertions, 8 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index 6536946..0d26562 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -13,9 +13,6 @@
 
 #define _WINSOCKAPI_		// prevents the inclusion of winsock.h
 
-#define RTP_SESSION_TIMEOUT 10
-#define RTP_WITH_SIGNALLING_SESSION_TIMEOUT (5*60)
-
 #include "Utils.h"
 #include "AudioCapture.h"
 #include "RtpSession.h"
@@ -970,7 +967,7 @@ void RtpSessions::ReportRtpPacket(RtpPacketInfoRef& rtpPacket)
 
 void RtpSessions::StopAll()
 {
-	time_t forceExpiryTime = time(NULL) + 2*RTP_WITH_SIGNALLING_SESSION_TIMEOUT;
+	time_t forceExpiryTime = time(NULL) + 2*DLLCONFIG.m_rtpSessionWithSignallingTimeoutSec;
 	Hoover(forceExpiryTime);
 }
 
@@ -989,11 +986,11 @@ void RtpSessions::Hoover(time_t now)
 		int timeoutSeconds = 0;
 		if(session->m_protocol == RtpSession::ProtRawRtp)
 		{
-			timeoutSeconds = RTP_SESSION_TIMEOUT;
+			timeoutSeconds = DLLCONFIG.m_rtpSessionTimeoutSec;
 		}
 		else
 		{
-			timeoutSeconds = RTP_WITH_SIGNALLING_SESSION_TIMEOUT;
+			timeoutSeconds = DLLCONFIG.m_rtpSessionWithSignallingTimeoutSec;
 		}
 		if((now - session->m_lastUpdated) > timeoutSeconds)
 		{
@@ -1014,7 +1011,7 @@ void RtpSessions::Hoover(time_t now)
 	for(pair = m_byCallId.begin(); pair != m_byCallId.end(); pair++)
 	{
 		RtpSessionRef session = pair->second;
-		if((now - session->m_lastUpdated) > RTP_WITH_SIGNALLING_SESSION_TIMEOUT)
+		if((now - session->m_lastUpdated) > DLLCONFIG.m_rtpSessionWithSignallingTimeoutSec)
 		{
 			toDismiss.push_back(session);
 		}
diff --git a/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp b/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
index eeab1b8..2b5b8f8 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
@@ -30,6 +30,8 @@ VoIpConfig::VoIpConfig()
 	m_pcapSocketBufferSize = 0;
 	m_pcapFastReplay = true;
 	m_pcapFastReplaySleepUsPerSec = 0;
+	m_rtpSessionTimeoutSec = 10;
+	m_rtpSessionWithSignallingTimeoutSec = 5*60;
 }
 
 void VoIpConfig::Define(Serializer* s)
@@ -49,7 +51,8 @@ void VoIpConfig::Define(Serializer* s)
 	s->IntValue("PcapFastReplaySleepUsPerSec", m_pcapFastReplaySleepUsPerSec);
 	s->BoolValue("SipDropIndirectInvite", m_sipDropIndirectInvite);
 	s->IntValue("PcapSocketBufferSize", m_pcapSocketBufferSize);
-
+	s->IntValue("RtpSessionTimeoutSec", m_rtpSessionTimeoutSec);
+	s->IntValue("RtpSessionWithSignallingTimeoutSec", m_rtpSessionWithSignallingTimeoutSec);
 }
 
 void VoIpConfig::Validate()
@@ -184,6 +187,18 @@ void VoIpConfig::Validate()
 		throw (exception);
 
 	}
+	if(m_rtpSessionTimeoutSec < 1)
+	{
+		CStdString exception;
+		exception.Format("VoIpConfig: RtpSessionTimeoutSec must be > 0 (currently:%d) please fix config.xml", m_rtpSessionTimeoutSec);
+		throw (exception);
+	}
+	if(m_rtpSessionWithSignallingTimeoutSec < 1)
+	{
+		CStdString exception;
+		exception.Format("VoIpConfig: RtpSessionWithSignallingTimeoutSec must be > 0 (currently:%d) please fix config.xml", m_rtpSessionWithSignallingTimeoutSec);
+		throw (exception);
+	}
 }
 
 bool VoIpConfig::IsPartOfLan(struct in_addr addr)
diff --git a/orkaudio/audiocaptureplugins/voip/VoIpConfig.h b/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
index 4ecd2c7..e2c08ee 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
+++ b/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
@@ -60,6 +60,9 @@ public:
 	int m_pcapSocketBufferSize;
 	bool m_pcapFastReplay;
 	int m_pcapFastReplaySleepUsPerSec;
+	int m_rtpSessionTimeoutSec;
+	int m_rtpSessionWithSignallingTimeoutSec;
+
 };
 
 //========================================