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

git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/1.8.15@375585 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6d9c5b42b6c137935a583feb7d4e5b5403dee232..e86e3d00e64e7199334548d8fe4740c1c32ad427 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -27497,6 +27497,10 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
                                } 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);
                                }
@@ -30037,6 +30041,18 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
                p->redircodecs = codecs;
                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)

diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 5165c41357ebd967f9a1af3d185346e5b2026415..c1263f075daf0f07b712b178f710f7edd475b66e 100644 (file)
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -359,9 +359,10 @@
 
 
 #define SIP_PAGE3_SNOM_AOC               (1 << 0)  /*!< DPG: Allow snom aoc messages */
+#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING  (1 << 1)  /*!< DP: Only send direct media reinvites on outgoing calls */
 
 #define SIP_PAGE3_FLAGS_TO_COPY \
-       (SIP_PAGE3_SNOM_AOC)
+       (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
 
 /*@}*/
 

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index bd5285517a7778b48aaab488ced97f79e9077c2c..4a1f76d1c3a25f0faac3ea8fb6772aaed17a546f 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -892,6 +892,13 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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.
                                 ; Will not work for video and cases where the callee sends