13 Commits (master)

Author	SHA1	Message	Date
Victor Seva	2bf464e5cb	Imported Upstream version 1.6.0~20150612~a2d214df36	11 years ago
Victor Seva	df9992620d	Imported Upstream version 1.6.0~20140516~eaa616	12 years ago
Stefan Sayer	26fdaa8eaf	b/f: stopping failed outbound calls ... applications which are doing outbound calls should implement onFailedOutboundCall if the behavior should be different than setting stopped. Reported by Andrei Samusenko	14 years ago
Stefan Sayer	bed6c2545a	b/f: in webconference app, handle session timer timeout and RTP timeout ... by leaving the conference	14 years ago
Stefan Sayer	b2b11301ff	webconference: participant ID / findParticipant	15 years ago
Stefan Sayer	f5846bfa27	webconference: roomDelete and time for roomCreate ... added config items (webconference.conf): private_rooms support_rooms_timeout implements bugs #0000048 and #0000049	15 years ago
Stefan Sayer	7dcb7e2abb	added OpenSSL linking exception to license	16 years ago
Stefan Sayer	90cccb3383	reworked SST + B2BUA implementation ... - B2BUA app with SIP Session Timer (SST) now can also use UPDATE (see session_refresh_method in sst_b2b.conf) - if re-INVITE is used, normal SDP OA (INVITE+SDP/200+SDP) is done using last established SDP (instead of delayed SDP negotiation and SDP ping-pong) - SDP is compared only after o= line (no SDP ping-pong with UAs which always increase SDP version)	16 years ago
Stefan Sayer	7fbe92bbd7	fix authentication on outgoing calls ... uac auth now resets the dialog status back to what it was before the failed request, if a request is retried with authentication.	16 years ago
Raphael Coeffic	dea103ae68	- make the AmSipDialog call the SIP event handler within AmSession, instead of the opposite. ... - WARNING: this code is not yet tested. git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1827 8eb893ce-cfd4-0310-b710-fb5ebe64c474	16 years ago
Stefan Sayer	c6d9ae3bbf	create RTP stream (AmSession::rtp_str) only if really needed ... 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	16 years ago
Stefan Sayer	f30989deff	mass setting Id svn property ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1048 8eb893ce-cfd4-0310-b710-fb5ebe64c474	18 years ago
Stefan Sayer	79526cdf4a	moved session related code to WebConferenceDialog.[cpp,h] ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@907 8eb893ce-cfd4-0310-b710-fb5ebe64c474	18 years ago