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Ensure entering T.38 passthrough does not cause an infinite loop

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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
This commit is contained in:
Kinsey Moore
2012-02-02 22:26:50 +00:00
parent c5fc58c3a0
commit ea4fa3227f
1 changed files with 8 additions and 0 deletions
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8
channels/chan_sip.c
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@@ -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 /* Ensure RTCP is enabled since it may be inactive
if we're coming back from a T.38 session */ if we're coming back from a T.38 session */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1); 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) { if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF); 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) { } else if (udptlportno > 0) {
if (debug) if (debug)
ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n"); 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 */ /* Silence RTCP while audio RTP is inactive */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0); ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
} else { } else {