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 309a21b4a4e5dac861637cee9588f1df9b41902e..9b8c5836d3e88b45334d036ffd5f64ec2c93dfb2 100644 (file)
--- 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 {