[asterisk-commits] kmoore: trunk r340972 - in /trunk: ./ channels/chan_sip.c res/res_rtp_asterisk.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Oct 14 15:51:23 CDT 2011
Author: kmoore
Date: Fri Oct 14 15:51:19 2011
New Revision: 340972
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=340972
Log:
Merged revisions 340971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................
r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
Merged revisions 340970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
Quiet RTCP Receiver Reports during fax transmission
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)
........
................
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/res/res_rtp_asterisk.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=340972&r1=340971&r2=340972
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Oct 14 15:51:19 2011
@@ -9127,6 +9127,9 @@
}
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) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -9143,6 +9146,8 @@
} else if (udptlportno > 0) {
if (debug)
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 {
ast_rtp_instance_stop(p->rtp);
if (debug)
Modified: trunk/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_rtp_asterisk.c?view=diff&rev=340972&r1=340971&r2=340972
==============================================================================
--- trunk/res/res_rtp_asterisk.c (original)
+++ trunk/res/res_rtp_asterisk.c Fri Oct 14 15:51:19 2011
@@ -2382,44 +2382,65 @@
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
if (property == AST_RTP_PROPERTY_RTCP) {
- if (rtp->rtcp) {
- ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance);
+ if (value) {
+ if (rtp->rtcp) {
+ ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance);
+ return;
+ }
+ /* Setup RTCP to be activated on the next RTP write */
+ if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) {
+ return;
+ }
+
+ /* Grab the IP address and port we are going to use */
+ ast_rtp_instance_get_local_address(instance, &rtp->rtcp->us);
+ ast_sockaddr_set_port(&rtp->rtcp->us,
+ ast_sockaddr_port(&rtp->rtcp->us) + 1);
+
+ if ((rtp->rtcp->s =
+ create_new_socket("RTCP",
+ ast_sockaddr_is_ipv4(&rtp->rtcp->us) ?
+ AF_INET :
+ ast_sockaddr_is_ipv6(&rtp->rtcp->us) ?
+ AF_INET6 : -1)) < 0) {
+ ast_debug(1, "Failed to create a new socket for RTCP on instance '%p'\n", instance);
+ ast_free(rtp->rtcp);
+ rtp->rtcp = NULL;
+ return;
+ }
+
+ /* Try to actually bind to the IP address and port we are going to use for RTCP, if this fails we have to bail out */
+ if (ast_bind(rtp->rtcp->s, &rtp->rtcp->us)) {
+ ast_debug(1, "Failed to setup RTCP on RTP instance '%p'\n", instance);
+ close(rtp->rtcp->s);
+ ast_free(rtp->rtcp);
+ rtp->rtcp = NULL;
+ return;
+ }
+
+ ast_debug(1, "Setup RTCP on RTP instance '%p'\n", instance);
+ rtp->rtcp->schedid = -1;
+
return;
- }
- if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) {
+ } 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;
}
-
- /* Grab the IP address and port we are going to use */
- ast_rtp_instance_get_local_address(instance, &rtp->rtcp->us);
- ast_sockaddr_set_port(&rtp->rtcp->us,
- ast_sockaddr_port(&rtp->rtcp->us) + 1);
-
- if ((rtp->rtcp->s =
- create_new_socket("RTCP",
- ast_sockaddr_is_ipv4(&rtp->rtcp->us) ?
- AF_INET :
- ast_sockaddr_is_ipv6(&rtp->rtcp->us) ?
- AF_INET6 : -1)) < 0) {
- ast_debug(1, "Failed to create a new socket for RTCP on instance '%p'\n", instance);
- ast_free(rtp->rtcp);
- rtp->rtcp = NULL;
- return;
- }
-
- /* Try to actually bind to the IP address and port we are going to use for RTCP, if this fails we have to bail out */
- if (ast_bind(rtp->rtcp->s, &rtp->rtcp->us)) {
- ast_debug(1, "Failed to setup RTCP on RTP instance '%p'\n", instance);
- close(rtp->rtcp->s);
- ast_free(rtp->rtcp);
- rtp->rtcp = NULL;
- return;
- }
-
- ast_debug(1, "Setup RTCP on RTP instance '%p'\n", instance);
- rtp->rtcp->schedid = -1;
-
- return;
}
return;
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