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
........

Merged revisions 353915 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 353916 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353917 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a46fad3..adc6a3b 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -9445,6 +9445,10 @@ 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);
@@ -9461,6 +9465,10 @@ 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 {