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path: root/orkaudio/audiocaptureplugins/voip/RtpSession.h
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2006-12-28SIP Session metadata can now be updated when a new RTP stream is seen that ↵Henri Herscher
matches one of the INVITE messages associated with the session. When this happens, a new tape message with stage=update is generated with the updated metadata. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@381 09dcff7a-b715-0410-9601-b79a96267cd0
2006-12-271. Subsequent SIP INVITES that could be associated to an existing session ↵Henri Herscher
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
2006-12-12RTP sessions now check the destination IP address of a packet to determine ↵Henri Herscher
whether it's side 1. This allows incoming PSTN calls transferred to various IP endpoints to be recorded as one session under two conditions: 1. There is no signalling for the second and subsequent segments and 2. the first RTP packet is sent from an internal endpoint to the PSTN gateway, see remark (1) in RtpSession.cpp. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@370 09dcff7a-b715-0410-9601-b79a96267cd0
2006-12-12Now possible to split RTP sessions based on timestamp and sequence number ↵Henri Herscher
discontinuity. This is disabled by default and can be enabled using the <RtpDiscontinuityDetect> config.xml boolean parameter. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@369 09dcff7a-b715-0410-9601-b79a96267cd0
2006-11-21Now logging whether a session has seen duplicate RTP traffic and the maximum ↵Henri Herscher
delta in sequence number seen during the capture (logged at session stop). git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@356 09dcff7a-b715-0410-9601-b79a96267cd0
2006-10-11There is now an orkaudio wide tracking ID for recording sessions. All ↵Henri Herscher
logging related to a recording session includes this tracking ID. The recording file name now uses this tracking ID instead of the "port" name which used to be IP+TcpPort of one endpoint. Also introduced the notion of OrkUid which is an unique identifier based on timestamp + tracking ID. OrkUid is not yet reported to orktrack. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@342 09dcff7a-b715-0410-9601-b79a96267cd0
2006-08-01Applying changeset 269 from 0.5 to trunk:Henri Herscher
Can now get local party info from the Skinny LineStatMessage. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@320 09dcff7a-b715-0410-9601-b79a96267cd0
2006-08-01Applying changeset 266 from 0.5 to trunk:Henri Herscher
Removed the unused old RTP mixer from the VoIp.dll git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@319 09dcff7a-b715-0410-9601-b79a96267cd0
2006-08-01Applying changeset 0.5 263:Henri Herscher
* Redundant RTP packets are now detected and discarded, on the basis of RTP timestamp. * RTP timestamp corrective delta now unsigned int in case it is bigger than 2^31, also using another boolean variable for sign. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@316 09dcff7a-b715-0410-9601-b79a96267cd0
2006-06-05RTP session capturePort is now allways aligned with ipAndPort so as to avoid ↵Henri Herscher
the possibility of having two concurrent RTP sessions for one OrkAudio capture port. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@246 09dcff7a-b715-0410-9601-b79a96267cd0
2006-05-26RTP timestamp corrective delta is now always applied because in some cases, ↵Henri Herscher
it produced garbled output on one side of the conversation. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@242 09dcff7a-b715-0410-9601-b79a96267cd0
2006-04-25Improved skinny signalling detection by taking OpenReceiveChannelAck Skinny ↵Henri Herscher
messages into account. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@222 09dcff7a-b715-0410-9601-b79a96267cd0
2006-04-14OrkAudio now reports local and remote IP addresses for each tape when the ↵Henri Herscher
voip plugin is used. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@217 09dcff7a-b715-0410-9601-b79a96267cd0
2006-04-08* Added allowed and blocked CIDR IP address rangesHenri Herscher
* Fixed SIP to: field detection issue * Fixed SIP multiple invite issue * replaced sleep by nanosleep * Improved logging git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@212 09dcff7a-b715-0410-9601-b79a96267cd0
2006-03-28Improved initializationHenri Herscher
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@202 09dcff7a-b715-0410-9601-b79a96267cd0
2006-02-28Skinny StartMedia is now associated to CallInfo based on IP address of the ↵Henri Herscher
endpoint when the StartMedia conference ID is null - additionally, SIP "INVITE" string and "BYE" string are now strcmp'ed for the full string (avoid SIP false positive). git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@187 09dcff7a-b715-0410-9601-b79a96267cd0
2006-02-03Duplicate SIP invites now disregarded so that sessions do not stop ↵Henri Herscher
spuriously. Also when a pcap file is replayed, all RTP sessions are now closed before the thread ends so as to avoid hanging RTP sessions. git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@156 09dcff7a-b715-0410-9601-b79a96267cd0
2005-12-14Improved SIP loggingHenri Herscher
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@110 09dcff7a-b715-0410-9601-b79a96267cd0
2005-12-14Added tracking ID to RTP sessionsHenri Herscher
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@108 09dcff7a-b715-0410-9601-b79a96267cd0
2005-12-09Can now handle when two different sessions are created for the same actual ↵Henri Herscher
bidirectional RTP session (observed on CallManager where each way has it's own callid/RTP timestamps) git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@102 09dcff7a-b715-0410-9601-b79a96267cd0
2005-12-08Oreka now supports mixing both sides into one file on Cisco CallManagerHenri Herscher
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@98 09dcff7a-b715-0410-9601-b79a96267cd0
2005-11-01Completed skinny supportHenri Herscher
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@25 09dcff7a-b715-0410-9601-b79a96267cd0
2005-10-31Groundwork for adding skinny supportHenri Herscher
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@24 09dcff7a-b715-0410-9601-b79a96267cd0
2005-10-28SipSession.cpp/.h become RtpSession.cpp/.hHenri Herscher
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@23 09dcff7a-b715-0410-9601-b79a96267cd0