[asterisk-commits] oej: branch oej/pinefrog-rtcp-1.8 r382723 - in /team/oej/pinefrog-rtcp-1.8/ch...

[SVN commits to the Asterisk project]

Author: oej
Date: Fri Mar  8 10:51:57 2013
New Revision: 382723

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=382723
Log:
Tell me more, tell me more, tell me more. (quote from Grease - the musical )

Modified:
    team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c
    team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c

Modified: team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c?view=diff&rev=382723&r1=382722&r2=382723
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c (original)
+++ team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c Fri Mar  8 10:51:57 2013
@@ -21621,12 +21621,12 @@
 {
 	/* Immediately stop RTP, VRTP and UDPTL as applicable */
 	if (p->rtp && !ast_rtp_instance_isactive(p->rtp)) {
+		sip_rtcp_report(p, p->rtp, SDP_AUDIO, TRUE);
 		ast_rtp_instance_stop(p->rtp);
-		sip_rtcp_report(p, p->rtp, SDP_AUDIO, TRUE);
 	}
 	if (p->vrtp && !ast_rtp_instance_isactive(p->vrtp)) {
+		sip_rtcp_report(p, p->vrtp, SDP_VIDEO, TRUE);
 		ast_rtp_instance_stop(p->vrtp);
-		sip_rtcp_report(p, p->vrtp, SDP_VIDEO, TRUE);
 	}
 	if (p->trtp && !ast_rtp_instance_isactive(p->trtp)) {
 		ast_rtp_instance_stop(p->trtp);

Modified: team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c?view=diff&rev=382723&r1=382722&r2=382723
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c (original)
+++ team/oej/pinefrog-rtcp-1.8/channels/sip/rtcp.c Fri Mar  8 10:51:57 2013
@@ -304,10 +304,10 @@
 	ast_log(LOG_DEBUG, "***** SENDING RTCP EVENT \n");
 
 	if (dialog->rtp && !ast_rtp_instance_isactive(dialog->rtp)) {
-		ast_log(LOG_DEBUG, "          ***** Activating RTCP report \n");
+		ast_debug(1, "          ***** Activating RTCP report \n");
 		sip_rtcp_report(dialog, dialog->rtp, SDP_AUDIO, FALSE);
 	} else {
-		ast_log(LOG_DEBUG, "          ***** NOT Activating RTCP report \n");
+		ast_debug(1, "          ***** NOT Activating RTCP report \n");
 	}
 	if (dialog->vrtp && !ast_rtp_instance_isactive(dialog->vrtp)) {
 		sip_rtcp_report(dialog, dialog->vrtp, SDP_VIDEO, FALSE);
@@ -318,8 +318,9 @@
 /*! \brief Activate RTCP events at start of call */
 void start_rtcp_events(struct sip_pvt *dialog, struct sched_context *sched)
 {
-	ast_log(LOG_DEBUG, "***** STARTING SENDING RTCP EVENT \n");
+	ast_debug(2, "***** STARTING SENDING RTCP EVENT \n");
 	if (!dialog->sip_cfg->rtcpevents || !dialog->sip_cfg->rtcptimer) {
+		ast_debug(2, "***** NOT SENDING RTCP EVENTS \n");
 		return;
 	}
 	/* Check if it's already active */