[asterisk-commits] file: branch 1.4 r112204 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Tue Apr  1 12:43:46 2008
New Revision: 112204

URL: http://svn.digium.com/view/asterisk?view=rev&rev=112204
Log:
Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered.
(closes issue #11823)
Reported by: SDamm

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=112204&r1=112203&r2=112204
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Apr  1 12:43:46 2008
@@ -4376,6 +4376,11 @@
 				ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
 			ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
 		}
+	}
+
+	/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
+	if (p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
+		fr = &ast_null_frame;
 	}
 
 	ast_mutex_unlock(&p->lock);