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.

Review: https://reviewboard.asterisk.org/r/1954



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -28025,6 +28025,10 @@ 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);
 				}
@@ -30706,6 +30710,18 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
 		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 (ast_channel_state(chan) != AST_STATE_UP) {     /* We are in early state */
 			if (p->do_history)

channels/sip/include/sip.h

@@ -368,9 +368,11 @@
 #define SIP_PAGE3_SRTP_TAG_32            (1 << 1)  /*!< DP: Use a 32bit auth tag in INVITE not 80bit */
 #define SIP_PAGE3_NAT_AUTO_RPORT         (1 << 2)  /*!< DGP: Set SIP_NAT_FORCE_RPORT when NAT is detected */
 #define SIP_PAGE3_NAT_AUTO_COMEDIA       (1 << 3)  /*!< DGP: Set SIP_PAGE2_SYMMETRICRTP when NAT is detected */
+#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING  (1 << 4)  /*!< 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_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA)
+	(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
+	 SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
 
 #define CHECK_AUTH_BUF_INITLEN   256
 

configs/sip.conf.sample

@@ -936,6 +936,13 @@ 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