[asterisk-commits] file: branch group/performance r399665 - /team/group/performance/channels/

[SVN commits to the Asterisk project]

Author: file
Date: Tue Sep 24 12:45:09 2013
New Revision: 399665

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=399665
Log:
When setting RTCP related variables only create and publish a single channel snapshot.

Modified:
    team/group/performance/channels/chan_sip.c

Modified: team/group/performance/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/performance/channels/chan_sip.c?view=diff&rev=399665&r1=399664&r2=399665
==============================================================================
--- team/group/performance/channels/chan_sip.c (original)
+++ team/group/performance/channels/chan_sip.c Tue Sep 24 12:45:09 2013
@@ -26516,6 +26516,10 @@
 		}
 	}
 
+	if ((p->rtp || p->vrtp || p->trtp) && p->owner) {
+		ast_channel_stage_snapshot(p->owner);
+	}
+
 	/* Get RTCP quality before end of call */
 	if (p->rtp) {
 		if (p->do_history) {
@@ -26578,6 +26582,10 @@
 		if (p->owner) {
 			pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
 		}
+	}
+
+	if ((p->rtp || p->vrtp || p->trtp) && p->owner) {
+		ast_channel_stage_snapshot_done(p->owner);
 	}
 
 	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */