[asterisk-commits] oej: branch oej/pinefrog-rtcp-1.8 r382708 - /team/oej/pinefrog-rtcp-1.8/res/

[SVN commits to the Asterisk project]

Author: oej
Date: Fri Mar  8 10:12:54 2013
New Revision: 382708

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=382708
Log:
Delay p2p bridge forward a tiny bit so we get statistics to work with 

Modified:
    team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c

Modified: team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c?view=diff&rev=382708&r1=382707&r2=382708
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c (original)
+++ team/oej/pinefrog-rtcp-1.8/res/res_rtp_asterisk.c Fri Mar  8 10:12:54 2013
@@ -2524,10 +2524,6 @@
 		}
 	}
 
-	/* If we are directly bridged to another instance send the audio directly out */
-	if (ast_rtp_instance_get_bridged(instance) && !bridge_p2p_rtp_write(instance, rtpheader, res, hdrlen)) {
-		return &ast_null_frame;
-	}
 
 	/* If the version is not what we expected by this point then just drop the packet */
 	if (version != 2) {
@@ -2570,6 +2566,12 @@
 
 	/* Schedule RTCP report transmissions if possible */
 	ast_rtcp_schedule(instance);
+
+	/* This needs to be after RTCP calculations to get more RTCP data */
+	/* If we are directly bridged to another instance send the audio directly out */
+	if (ast_rtp_instance_get_bridged(instance) && !bridge_p2p_rtp_write(instance, rtpheader, res, hdrlen)) {
+		return &ast_null_frame;
+	}
 
 	/* Remove any padding bytes that may be present */
 	if (padding) {