[asterisk-commits] oej: branch oej/pinefrog-rtcp-1.8 r382681 - /team/oej/pinefrog-rtcp-1.8/chann...

[SVN commits to the Asterisk project]

Author: oej
Date: Fri Mar  8 05:02:33 2013
New Revision: 382681

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=382681
Log:
Why oh why do you stop all the RTP flows because you are simply on hold? RTP keepalives and RTCP still needs to flow,
you little Asterisk rascal. Let's try this little surgical change and see what happens.

Modified:
    team/oej/pinefrog-rtcp-1.8/channels/chan_sip.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=382681&r1=382680&r2=382681
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c (original)
+++ team/oej/pinefrog-rtcp-1.8/channels/chan_sip.c Fri Mar  8 05:02:33 2013
@@ -9884,11 +9884,12 @@
 		ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
 				       S_OR(p->mohsuggest, NULL),
 				       !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
-		if (sendonly) {
-			ast_rtp_instance_stop(p->rtp);
-			/* RTCP needs to go ahead, even if we're on hold!!! */
-			/* OEJ this is a BUG to fix. Now. */
-		}
+		//
+		// We should only stop at port = 0 or BYE /End of call or stream (like a transfer)
+		//if (sendonly == 2) {
+			//ast_rtp_instance_stop(p->rtp);
+			///* RTCP needs to go ahead, even if we're on hold!!! */
+		//}
 		/* Activate a re-invite */
 		ast_queue_frame(p->owner, &ast_null_frame);
 		change_hold_state(p, req, TRUE, sendonly);