[asterisk-commits] kmoore: branch 1.8 r353915 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: kmoore
Date: Thu Feb  2 16:26:50 2012
New Revision: 353915

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=353915
Log:
Ensure entering T.38 passthrough does not cause an infinite loop

After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is
shut down and removed. If the descriptor happened to have data ready when the
removal occured then Asterisk would go into an infinite loop trying to read
data that it can never actually access. This change disables the audio RTCP
file descriptor for the duration of the T.38 transaction.

(closes issue ASTERISK-18951)
Reported-by: Kristijan Vrban

Modified:
    branches/1.8/channels/chan_sip.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=353915&r1=353914&r2=353915
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Thu Feb  2 16:26:50 2012
@@ -9216,6 +9216,10 @@
 			/* Ensure RTCP is enabled since it may be inactive
 			   if we're coming back from a T.38 session */
 			ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
+			/* Ensure audio RTCP reads are enabled */
+			if (p->owner) {
+				ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
+			}
 
 			if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
 				ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -9232,6 +9236,10 @@
 		} else if (udptlportno > 0) {
 			if (debug)
 				ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
+			/* Prevent audio RTCP reads */
+			if (p->owner) {
+				ast_channel_set_fd(p->owner, 1, -1);
+			}
 			/* Silence RTCP while audio RTP is inactive */
 			ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
 		} else {