now cast all strutct in_addr.s_addr to unsigned int before comparing.

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@206 09dcff7a-b715-0410-9601-b79a96267cd0

2 files changed, 12 insertions, 6 deletions

diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index 28e1c37..0c194e3 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -146,13 +146,13 @@ void RtpSession::ProcessMetadataSip(RtpPacketInfoRef& rtpPacket)
 	bool done = false;
 
 	// work out invitee IP address
-	if(rtpPacket->m_sourceIp.s_addr == m_invitorIp.s_addr)
+	if((unsigned int)rtpPacket->m_sourceIp.s_addr == (unsigned int)m_invitorIp.s_addr)
 	{
 		m_inviteeIp = rtpPacket->m_destIp;
 		m_inviteeTcpPort = rtpPacket->m_destPort;
 		m_invitorTcpPort = rtpPacket->m_sourcePort;
 	}
-	else if(rtpPacket->m_destIp.s_addr == m_invitorIp.s_addr)
+	else if((unsigned int)rtpPacket->m_destIp.s_addr == (unsigned int)m_invitorIp.s_addr)
 	{
 		m_inviteeIp = rtpPacket->m_sourceIp;
 		m_inviteeTcpPort = rtpPacket->m_sourcePort;
@@ -275,7 +275,7 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 	}
 	else
 	{
-		if(rtpPacket->m_sourceIp.s_addr == m_lastRtpPacketSide1->m_sourceIp.s_addr)
+		if((unsigned int)rtpPacket->m_sourceIp.s_addr == (unsigned int)m_lastRtpPacketSide1->m_sourceIp.s_addr)
 		{
 			m_lastRtpPacketSide1 = rtpPacket;
 			channel = 1;
@@ -526,7 +526,7 @@ void RtpSessions::ReportSkinnyStartMediaTransmission(SkStartMediaTransmissionStr
 		{
 			RtpSessionRef tmpSession = pair->second;
 
-			if(tmpSession->m_endPointIp.s_addr == ipHeader->ip_dest.s_addr)
+			if((unsigned int)tmpSession->m_endPointIp.s_addr == (unsigned int)ipHeader->ip_dest.s_addr)
 			{
 				if(tmpSession->m_ipAndPort.size() == 0)
 				{
diff --git a/orkaudio/audiocaptureplugins/voip/VoIp.cpp b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
index f9124c8..96e7349 100644
--- a/orkaudio/audiocaptureplugins/voip/VoIp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/VoIp.cpp
@@ -222,6 +222,7 @@ bool TrySipBye(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, U
 bool TrySipInvite(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, UdpHeaderStruct* udpHeader, u_char* udpPayload)
 {
 	bool result = false;
+	bool drop = false;
 	if (memcmp("INVITE", (void*)udpPayload, 6) == 0)
 	{
 		result = true;
@@ -298,16 +299,21 @@ bool TrySipInvite(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader
 				if(ACE_OS::inet_aton((PCSTR)connectionAddress, &fromIp))
 				{
 					info->m_fromIp = fromIp;
+
+					//if((unsigned int)fromIp.s_addr != (unsigned int)ipHeader->ip_src.s_addr)
+					//{
+					//	drop =true;
+					//}
 				}
 			}
 		}
-		if(info->m_fromIp.s_addr == 0)
+		if((unsigned int)info->m_fromIp.s_addr == 0)
 		{
 			// In case connection address could not be extracted, use SIP invite sender IP address
 			info->m_fromIp = ipHeader->ip_src;
 		}
 
-		if(info->m_fromRtpPort.size() && info->m_from.size() && info->m_to.size() && info->m_callId.size())
+		if(drop ==false && info->m_fromRtpPort.size() && info->m_from.size() && info->m_to.size() && info->m_callId.size())
 		{
 			RtpSessionsSingleton::instance()->ReportSipInvite(info);
 		}