[svn-commits] irroot: branch irroot/t38gateway-1.8 r341073 - in /team/irroot/t38gateway-1.8...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Oct 16 11:53:07 CDT 2011
Author: irroot
Date: Sun Oct 16 11:52:59 2011
New Revision: 341073
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=341073
Log:
Multiple revisions 340970,341022
........
r340970 | kmoore | 2011-10-14 22:49:39 +0200 (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)
........
r341022 | kpfleming | 2011-10-14 23:36:06 +0200 (Fri, 14 Oct 2011) | 10 lines
Change the internal name of the menuselect options that are used to control
whether modules are embedded or not; using just the bare category name led to
accidentally enabling these options when users used the wrong "--enable"
operation on the menuselect command line.
Now the internal option names are prefixed with "EMBED_", so they won't be
the same as the name of the category containing the modules they control
the embedding of.
........
Merged revisions 340970,341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Modified:
team/irroot/t38gateway-1.8/ (props changed)
team/irroot/t38gateway-1.8/Makefile.moddir_rules
team/irroot/t38gateway-1.8/build_tools/embed_modules.xml
team/irroot/t38gateway-1.8/channels/chan_sip.c
team/irroot/t38gateway-1.8/res/res_rtp_asterisk.c
Propchange: team/irroot/t38gateway-1.8/
------------------------------------------------------------------------------
automerge = *
Propchange: team/irroot/t38gateway-1.8/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Oct 16 11:52:59 2011
@@ -1,1 +1,1 @@
-/branches/1.8:1-340879
+/branches/1.8:1-341072
Modified: team/irroot/t38gateway-1.8/Makefile.moddir_rules
URL: http://svnview.digium.com/svn/asterisk/team/irroot/t38gateway-1.8/Makefile.moddir_rules?view=diff&rev=341073&r1=341072&r2=341073
==============================================================================
--- team/irroot/t38gateway-1.8/Makefile.moddir_rules (original)
+++ team/irroot/t38gateway-1.8/Makefile.moddir_rules Sun Oct 16 11:52:59 2011
@@ -1,6 +1,6 @@
#
# Asterisk -- A telephony toolkit for Linux.
-#
+#
# Makefile rules for subdirectories containing modules
#
# Copyright (C) 2006, Digium, Inc.
@@ -46,7 +46,7 @@
C_MODS:=$(filter-out $(MENUSELECT_$(MENUSELECT_CATEGORY)),$(ALL_C_MODS))
CC_MODS:=$(filter-out $(MENUSELECT_$(MENUSELECT_CATEGORY)),$(ALL_CC_MODS))
-ifneq ($(findstring $(MENUSELECT_CATEGORY),$(MENUSELECT_EMBED)),)
+ifneq ($(findstring EMBED_$(MENUSELECT_CATEGORY),$(MENUSELECT_EMBED)),)
EMBEDDED_MODS:=$(C_MODS) $(CC_MODS)
else
LOADABLE_MODS:=$(C_MODS) $(CC_MODS)
Modified: team/irroot/t38gateway-1.8/build_tools/embed_modules.xml
URL: http://svnview.digium.com/svn/asterisk/team/irroot/t38gateway-1.8/build_tools/embed_modules.xml?view=diff&rev=341073&r1=341072&r2=341073
==============================================================================
--- team/irroot/t38gateway-1.8/build_tools/embed_modules.xml (original)
+++ team/irroot/t38gateway-1.8/build_tools/embed_modules.xml Sun Oct 16 11:52:59 2011
@@ -1,45 +1,45 @@
<category name="MENUSELECT_EMBED" displayname="Module Embedding" positive_output="yes" remove_on_change="main/asterisk">
- <member name="ADDONS" displayname="Add-ons" remove_on_change="addons/*.o addons/*.oo">
+ <member name="EMBED_ADDONS" displayname="Add-ons" remove_on_change="addons/*.o addons/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="APPS" displayname="Applications" remove_on_change="apps/*.o apps/*.oo">
+ <member name="EMBED_APPS" displayname="Applications" remove_on_change="apps/*.o apps/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="BRIDGES" displayname="Bridging Technologies" remove_on_change="bridges/*.o bridges/*.oo">
+ <member name="EMBED_BRIDGES" displayname="Bridging Technologies" remove_on_change="bridges/*.o bridges/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="CDR" displayname="Call Detail Recording" remove_on_change="cdr/*.o cdr/*.oo">
+ <member name="EMBED_CDR" displayname="Call Detail Recording" remove_on_change="cdr/*.o cdr/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="CHANNELS" displayname="Channels" remove_on_change="channels/*.o channels/*/*.o channels/*.oo channels/*/*.oo">
+ <member name="EMBED_CHANNELS" displayname="Channels" remove_on_change="channels/*.o channels/*/*.o channels/*.oo channels/*/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="CODECS" displayname="Coders/Decoders" remove_on_change="codecs/*.o codecs/*/*.o codecs/*/*/*.o codecs/*/*.a codecs/*/*/*.a codecs/*.oo codecs/*/*.oo">
+ <member name="EMBED_CODECS" displayname="Coders/Decoders" remove_on_change="codecs/*.o codecs/*/*.o codecs/*/*/*.o codecs/*/*.a codecs/*/*/*.a codecs/*.oo codecs/*/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="FORMATS" displayname="File Formats" remove_on_change="formats/*.o formats/*.oo">
+ <member name="EMBED_FORMATS" displayname="File Formats" remove_on_change="formats/*.o formats/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="FUNCS" displayname="Dialplan Functions" remove_on_change="funcs/*.o funcs/*.oo">
+ <member name="EMBED_FUNCS" displayname="Dialplan Functions" remove_on_change="funcs/*.o funcs/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="PBX" displayname="PBX Functionality" remove_on_change="pbx/*.o pbx/*/*.o pbx/*.oo pbx/*/*.oo">
+ <member name="EMBED_PBX" displayname="PBX Functionality" remove_on_change="pbx/*.o pbx/*/*.o pbx/*.oo pbx/*/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="RES" displayname="Resource Modules" remove_on_change="res/*.o res/*/*.o res/*.oo res/*/*.oo">
+ <member name="EMBED_RES" displayname="Resource Modules" remove_on_change="res/*.o res/*/*.o res/*.oo res/*/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
- <member name="TEST" displayname="Test Modules" remove_on_change="tests/*.o tests/*.oo">
+ <member name="EMBED_TEST" displayname="Test Modules" remove_on_change="tests/*.o tests/*.oo">
<depend>gnu_ld</depend>
<support_level>extended</support_level>
</member>
Modified: team/irroot/t38gateway-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/irroot/t38gateway-1.8/channels/chan_sip.c?view=diff&rev=341073&r1=341072&r2=341073
==============================================================================
--- team/irroot/t38gateway-1.8/channels/chan_sip.c (original)
+++ team/irroot/t38gateway-1.8/channels/chan_sip.c Sun Oct 16 11:52:59 2011
@@ -8932,6 +8932,9 @@
}
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);
@@ -8948,6 +8951,8 @@
} 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: team/irroot/t38gateway-1.8/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/irroot/t38gateway-1.8/res/res_rtp_asterisk.c?view=diff&rev=341073&r1=341072&r2=341073
==============================================================================
--- team/irroot/t38gateway-1.8/res/res_rtp_asterisk.c (original)
+++ team/irroot/t38gateway-1.8/res/res_rtp_asterisk.c Sun Oct 16 11:52:59 2011
@@ -2381,44 +2381,65 @@
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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