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
to use it, set USE_THREADPOOL in Makefile.defs and configure thread pool
size with session_processor_threads= parameter in sems.conf :
+# compile with session thread pool support?
+# use this for very high concurrent call count
+# applications (e.g. for signaling only)
+# if compiled with thread pool, there will be a
+# thread pool of configurable size processing the
+# signaling and application logic of the calls.
+# if compiled without thread pool support, every
+# session will have its own thread.
+#
+#USE_THREADPOOL = yes
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1782 8eb893ce-cfd4-0310-b710-fb5ebe64c474
- static library sip_stack.a (with dependencies on the core).
- removed AmServer and moved SipCtrlInterface into the core directory.
- TODO:
- CMake support to make core/sip/sip_stack.a and link against it in the core.
- merge AmSipRequest/AmSipReply and sip_msg structures.
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1758 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
- removed AmSIPEventHandler (its functionalities are now in AmEventDispatcher).
- added possibility for each plug-in to receive out-of-dialog messages (any kind).
- added possibility to handle dialogs without creating a session (=AmSession).
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1006 8eb893ce-cfd4-0310-b710-fb5ebe64c474
o moved adding Max-Forwards from ctrl plugins to AmSipDialog
o moved adding User-Agent from ctrl plugins to AmSipDialog
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@760 8eb893ce-cfd4-0310-b710-fb5ebe64c474
in the RTP stream. It is possible to disable this new behaviour using
single_codec_in_ok parameter in the sems.conf.
Developed by: Sippy Software, Inc.
Sponsored by: Digifonica Canada Limited
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@279 8eb893ce-cfd4-0310-b710-fb5ebe64c474
* changed apps that use session specific params to use P-Iptel-Param by default
* added getSessionParam function to ivr
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@260 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
* new methods onSipRequest/onSipReply, the whole AmSipReply/AmSipRequest structure is passed to python
SipDialog
* state is changed only on reply to INVITE, not other requests
* made pin_collect work
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@189 8eb893ce-cfd4-0310-b710-fb5ebe64c474
Stefan Sayer
Raphael Coeffic
Richard Newman
Maxim Sobolev