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Quiet RTCP Receiver Reports during fax transmission

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RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
code was added to support the bug fix.

(closes issue ASTERISK-18400)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kinsey Moore
2011-10-14 20:49:39 +00:00
parent 927336fe2f
commit 0fa2f5914e
2 changed files with 62 additions and 36 deletions
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5
channels/chan_sip.c
View File

@@ -8928,6 +8928,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} }
ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp); ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
/* 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);
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);
@@ -8944,6 +8947,8 @@ 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");
/* Silence RTCP while audio RTP is inactive */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
} else { } else {
ast_rtp_instance_stop(p->rtp); ast_rtp_instance_stop(p->rtp);
if (debug) if (debug)

21
res/res_rtp_asterisk.c
View File

@@ -2381,10 +2381,12 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance); struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
if (property == AST_RTP_PROPERTY_RTCP) { if (property == AST_RTP_PROPERTY_RTCP) {
if (value) {
if (rtp->rtcp) { if (rtp->rtcp) {
ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance); ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance);
return; return;
} }
/* Setup RTCP to be activated on the next RTP write */
if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) { if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) {
return; return;
} }
@@ -2419,6 +2421,25 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
rtp->rtcp->schedid = -1; rtp->rtcp->schedid = -1;
return; return;
} else {
if (rtp->rtcp) {
if (rtp->rtcp->schedid > 0) {
if (!ast_sched_del(rtp->sched, rtp->rtcp->schedid)) {
/* Successfully cancelled scheduler entry. */
ao2_ref(instance, -1);
} else {
/* Unable to cancel scheduler entry */
ast_debug(1, "Failed to tear down RTCP on RTP instance '%p'\n", instance);
return;
}
rtp->rtcp->schedid = -1;
}
close(rtp->rtcp->s);
ast_free(rtp->rtcp);
rtp->rtcp = NULL;
}
return;
}
} }
return; return;