Added a boolean VoIP plugin configuration option, RtpReportDtmf, to control whether to report or not to report RTP events

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@497 09dcff7a-b715-0410-9601-b79a96267cd0

4 files changed, 36 insertions, 23 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index b133317..c52d9b6 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -596,12 +596,18 @@ bool RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 
 	if(m_protocol == ProtSip)
 	{
-		/* Check if this is a telephone-event */
-		if(rtpPacket->m_payloadType == StringToInt(m_telephoneEventPayloadType))
+		if(DLLCONFIG.m_rtpReportDtmf)
 		{
-			// This is a telephone-event
-			HandleRtpEvent(rtpPacket);
-			return true;
+			/* Check if this is a telephone-event */
+			if(m_telephoneEventPayloadType.CompareNoCase("UNDEFINED") != 0)
+			{
+				if(rtpPacket->m_payloadType == StringToInt(m_telephoneEventPayloadType))
+				{
+					// This is a telephone-event
+					HandleRtpEvent(rtpPacket);
+					return true;
+				}
+			}
 		}
 	}
 
@@ -1844,6 +1850,7 @@ SipInviteInfo::SipInviteInfo()
 	m_fromRtpIp.s_addr = 0;
 	m_validated = false;
 	m_attrSendonly = false;
+	m_telephoneEventPayloadType = "UNDEFINED";
 }
 
 void SipInviteInfo::ToString(CStdString& string)
diff --git a/orkaudio/audiocaptureplugins/voip/VoIp.cpp b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
index 6f3b81f..4790c7c 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
@@ -1641,31 +1641,34 @@ bool TrySipInvite(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader
 			}
 		}
 
-		if(rtpmapAttribute)
+		if(DLLCONFIG.m_rtpReportDtmf)
 		{
-			CStdString rtpPayloadType, nextToken;
-			char *nextStep = NULL;
-
-			while(rtpmapAttribute && rtpmapAttribute < sipEnd)
+			if(rtpmapAttribute)
 			{
-				GrabTokenSkipLeadingWhitespaces(rtpmapAttribute, sipEnd, rtpPayloadType);
-				nextToken.Format("%s ", rtpPayloadType);
-				nextStep = memFindAfter((char*)nextToken.c_str(), rtpmapAttribute, sipEnd);
+				CStdString rtpPayloadType, nextToken;
+				char *nextStep = NULL;
 
-				/* We need our "nextStep" to contain at least the length
-				 * of the string "telephone-event", 15 characters */
-				if(nextStep && ((sipEnd - nextStep) >= 15))
+				while(rtpmapAttribute && rtpmapAttribute < sipEnd)
 				{
-					if(ACE_OS::strncasecmp(nextStep, "telephone-event", 15) == 0)
+					GrabTokenSkipLeadingWhitespaces(rtpmapAttribute, sipEnd, rtpPayloadType);
+					nextToken.Format("%s ", rtpPayloadType);
+					nextStep = memFindAfter((char*)nextToken.c_str(), rtpmapAttribute, sipEnd);
+
+					/* We need our "nextStep" to contain at least the length
+					 * of the string "telephone-event", 15 characters */
+					if(nextStep && ((sipEnd - nextStep) >= 15))
 					{
-						/* Our DTMF packets are indicated using
-						 * the payload type rtpPayloadType */
-						info->m_telephoneEventPayloadType = rtpPayloadType;
-						break;
+						if(ACE_OS::strncasecmp(nextStep, "telephone-event", 15) == 0)
+						{
+							/* Our DTMF packets are indicated using
+							 * the payload type rtpPayloadType */
+							info->m_telephoneEventPayloadType = rtpPayloadType;
+							break;
+						}
 					}
-				}
 
-				rtpmapAttribute = memFindAfter("\na=rtpmap:", rtpmapAttribute, sipEnd);
+					rtpmapAttribute = memFindAfter("\na=rtpmap:", rtpmapAttribute, sipEnd);
+				}
 			}
 		}
 
diff --git a/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp b/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
index 89ab57b..d9f299e 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIpConfig.cpp
@@ -33,6 +33,7 @@ VoIpConfig::VoIpConfig()
 	m_rtpSessionTimeoutSec = 10;
 	m_rtpSessionWithSignallingTimeoutSec = 5*60;
 	m_rtpSessionOnHoldTimeOutSec = 1800;
+	m_rtpReportDtmf = false;
 	m_pcapTest= false;
 	m_rtpDiscontinuityDetect = false;
 	m_rtpDiscontinuityMinSeqDelta = 1000;
@@ -76,6 +77,7 @@ void VoIpConfig::Define(Serializer* s)
 	s->IntValue("RtpSessionTimeoutSec", m_rtpSessionTimeoutSec);
 	s->IntValue("RtpSessionWithSignallingTimeoutSec", m_rtpSessionWithSignallingTimeoutSec);
 	s->IntValue("RtpSessionOnHoldTimeOutSec", m_rtpSessionOnHoldTimeOutSec);
+	s->BoolValue("RtpReportDtmf", m_rtpReportDtmf);
 	s->BoolValue("PcapTest", m_pcapTest);
 	s->StringValue("PcapFilter", m_pcapFilter);
 	s->BoolValue("RtpDiscontinuityDetect", m_rtpDiscontinuityDetect);
diff --git a/orkaudio/audiocaptureplugins/voip/VoIpConfig.h b/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
index c1959a4..521e0ac 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
+++ b/orkaudio/audiocaptureplugins/voip/VoIpConfig.h
@@ -71,6 +71,7 @@ public:
 	bool m_rtpDiscontinuityDetect;
 	int  m_rtpDiscontinuityMinSeqDelta;
 	bool m_rtpDetectOnOddPorts;
+	bool m_rtpReportDtmf;
 	bool m_iax2Support;
 	bool m_sipOverTcpSupport;
 	bool m_sipLogFailedCalls;