replace AmSession::rtp_str with AmSession::RTPStream() in your app.
for example for pure signaling B2B calls, no RTP stream instance is
created, which saves a lot of memory (especially because of
the RTP receive buffer)
ref r30371 r30372
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1784 8eb893ce-cfd4-0310-b710-fb5ebe64c474
- active sessions and other event receivers
get a SystemEvent::ServerShutdown,
default behaviour of AmSession is setStopped()
- session container waits for all sessions to be ended
- signaling server, rtp receiver, media processor, event dispatcher
are stopped and deleted
based on a patch by Rui Jin Zheng rjzheng at boronetworks dot com
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1087 8eb893ce-cfd4-0310-b710-fb5ebe64c474
- on RTP timeout, AmSession::onRtpTimeout is called
- session can be removed from media processor (detached)
by AmMediaProcessor::removeSession
- removing session from MediaProcessor and clearing audio
is AmMediaProcessor::clearSession (default action on
RTP error)
- added detach from media processor to conf_auth and ann_b2b
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@209 8eb893ce-cfd4-0310-b710-fb5ebe64c474
Victor Seva
Stefan Sayer
Stefan Sayer
Raphael Coeffic