From: Kinsey Moore <kmoore@digium.com>
Date: Thu, 2 Feb 2012 22:26:50 +0000 (+0000)
Subject: Ensure entering T.38 passthrough does not cause an infinite loop
X-Git-Tag: 1.8.11.0-rc1~64
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=ea4fa3227f65b94599f69b9a60a85291569ffaee;p=thirdparty%2Fasterisk.git

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 309a21b4a4..9b8c5836d3 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -9216,6 +9216,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			/* 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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		} 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 {