[asterisk-commits] mmichelson: branch 10 r370619 - in /branches/10: ./ channels/ channels/sip/in...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 31 10:32:01 CDT 2012
Author: mmichelson
Date: Tue Jul 31 10:31:57 2012
New Revision: 370619
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370619
Log:
Help mitigate potential reinvite glare scenarios.
When Asterisk servers are set up back-to-back, and
direct media is to be used betweeen endpoints, it is
fairly common for the two Asterisk servers to send
direct media reinvites to each other simultaneously.
This results in 491s and ACKs being exchanged between
the servers. While the media eventually gets set up
properly, the problem is that there can be a noticeable
delay for the streams to stabilize.
This patch adds a new directmedia option called "outgoing".
With this set, an immediate direct media reinvite will only
be sent if the call direction is outgoing. For incoming
dialogs, an immediate direct media reinvite will not be sent,
but further "reactionary" direct media reinvites may be sent.
For those who are having some deja vu, that's because this
patch was originally committed to trunk since there is a
new configuration option added. After seeing a bug report
about audio being slow to set up on SIP calls, it became
apparent that this patch would be the best solution for
resolving the issue. The patch is unintrusive and will
have no effect unless the option is explicitly enabled.
(closes issue AST-896)
reported by Thomas Arimont
(closes issue ASTERISK-19857)
reported by Matt Jordan
........
Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Modified:
branches/10/ (props changed)
branches/10/channels/chan_sip.c
branches/10/channels/sip/include/sip.h
branches/10/configs/sip.conf.sample
Propchange: branches/10/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: branches/10/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/10/channels/chan_sip.c?view=diff&rev=370619&r1=370618&r2=370619
==============================================================================
--- branches/10/channels/chan_sip.c (original)
+++ branches/10/channels/chan_sip.c Tue Jul 31 10:31:57 2012
@@ -27901,6 +27901,10 @@
} else if (!strcasecmp(word, "nonat")) {
ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT);
+ } else if (!strcasecmp(word, "outgoing")) {
+ ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
+ ast_set_flag(&mask[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
+ ast_set_flag(&flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
} else {
ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno);
}
@@ -30473,6 +30477,18 @@
ast_format_cap_copy(p->redircaps, cap);
changed = 1;
}
+
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING) && !p->outgoing_call) {
+ /* We only wish to withhold sending the initial direct media reinvite on the incoming dialog.
+ * Further direct media reinvites beyond the initial should be sent. In order to allow further
+ * direct media reinvites to be sent, we clear this flag.
+ */
+ ast_clear_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
+ sip_pvt_unlock(p);
+ ast_channel_unlock(chan);
+ return 0;
+ }
+
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
if (chan->_state != AST_STATE_UP) { /* We are in early state */
if (p->do_history)
Modified: branches/10/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/branches/10/channels/sip/include/sip.h?view=diff&rev=370619&r1=370618&r2=370619
==============================================================================
--- branches/10/channels/sip/include/sip.h (original)
+++ branches/10/channels/sip/include/sip.h Tue Jul 31 10:31:57 2012
@@ -360,9 +360,10 @@
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
#define SIP_PAGE3_SRTP_TAG_32 (1 << 1) /*!< DP: Use a 32bit auth tag in INVITE not 80bit */
+#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 2) /*!< DP: Only send direct media reinvites on outgoing calls */
#define SIP_PAGE3_FLAGS_TO_COPY \
- (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32)
+ (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
#define CHECK_AUTH_BUF_INITLEN 256
Modified: branches/10/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/10/configs/sip.conf.sample?view=diff&rev=370619&r1=370618&r2=370619
==============================================================================
--- branches/10/configs/sip.conf.sample (original)
+++ branches/10/configs/sip.conf.sample Tue Jul 31 10:31:57 2012
@@ -887,6 +887,13 @@
;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
; instead of INVITE. This can be combined with 'nonat', as
; 'directmedia=update,nonat'. It implies 'yes'.
+
+;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
+ ; reinvite on an incoming call leg. This option is useful when
+ ; peered with another SIP user agent that is known to send
+ ; immediate direct media reinvites upon call establishment. Setting
+ ; the option in this situation helps to prevent potential glares.
+ ; Setting this option implies 'yes'.
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
; the call directly with media peer-2-peer without re-invites.
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