[svn-commits] rizzo: branch rizzo/astobj2 r57941 - /team/rizzo/astobj2/channels/chan_sip.c

svn-commits at lists.digium.com svn-commits at lists.digium.com
Mon Mar 5 12:57:54 MST 2007


Author: rizzo
Date: Mon Mar  5 13:57:53 2007
New Revision: 57941

URL: http://svn.digium.com/view/asterisk?view=rev&rev=57941
Log:
merge from trunk up to svn 53125

Modified:
    team/rizzo/astobj2/channels/chan_sip.c

Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=57941&r1=57940&r2=57941
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Mon Mar  5 13:57:53 2007
@@ -531,6 +531,7 @@
 static struct ast_codec_pref default_prefs;		/*!< Default codec prefs */
 
 /* Global settings only apply to the channel */
+static int global_directrtpsetup;       /*!< Enable support for Direct RTP setup (no re-invites) */
 static int global_limitonpeers;		/*!< Match call limit on peers only */
 static int global_rtautoclear;
 static int global_notifyringing;	/*!< Send notifications on ringing */
@@ -3240,6 +3241,7 @@
 
 	p->callingpres = ast->cid.cid_pres;
 	p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
+	p->jointnoncodeccapability = p->noncodeccapability;
 
 	/* If there are no audio formats left to offer, punt */
 	if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
@@ -5664,6 +5666,9 @@
 		if (newnoncodeccapability & AST_RTP_DTMF) {
 			/* XXX Would it be reasonable to drop the DSP at this point? XXX */
 			ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
+			/* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
+  	                ast_rtp_setdtmf(p->rtp, 1);
+  	                ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		} else {
 			ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
 		}
@@ -9965,7 +9970,7 @@
 			snprintf(ilimits, sizeof(ilimits), "%d", iterator->call_limit);
 		else 
 			ast_copy_string(ilimits, "N/A", sizeof(ilimits));
-		snprintf(iused, sizeof(iused), "%d/%d", iterator->inUse, iterator->inRinging);
+		snprintf(iused, sizeof(iused), "%d/%d/%d", iterator->inUse, iterator->inRinging, iterator->onHold);
 		if (showall || iterator->call_limit)
 			ast_cli(fd, FORMAT2, iterator->name, iused, ilimits);
 		ASTOBJ_UNLOCK(iterator);
@@ -10910,6 +10915,7 @@
 	ast_cli(fd, "  Realm. auth:            %s\n", authl ? "Yes": "No");
  	ast_cli(fd, "  Always auth rejects:    %s\n", global_alwaysauthreject ? "Yes" : "No");
 	ast_cli(fd, "  Call limit peers only:  %s\n", global_limitonpeers ? "Yes" : "No");
+	ast_cli(fd, "  Direct RTP setup:       %s\n", global_directrtpsetup ? "Yes" : "No");
 	ast_cli(fd, "  User Agent:             %s\n", global_useragent);
 	ast_cli(fd, "  MWI checking interval:  %d secs\n", global_mwitime);
 	ast_cli(fd, "  Reg. context:           %s\n", S_OR(global_regcontext, "(not set)"));
@@ -16937,6 +16943,7 @@
 	global_notifyringing = DEFAULT_NOTIFYRINGING;
 	global_limitonpeers = FALSE;		/*!< Match call limit on peers only */
 	global_notifyhold = FALSE;		/*!< Keep track of hold status for a peer */
+	global_directrtpsetup = FALSE;          /* Experimental feature, disabled by default */
 	global_alwaysauthreject = 0;
 	global_allowsubscribe = FALSE;
 	snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ASTERISK_VERSION);
@@ -17027,6 +17034,7 @@
 		M_BOOL("compactheaders", compactheaders)
 		M_STR("notifymimetype", default_notifymime)
 		M_BOOL("limitonpeers", global_limitonpeers)
+		M_BOOL("directrtpsetup", global_directrtpsetup)
 		M_BOOL("notifyringing", global_notifyringing)
 		M_BOOL("notifyhold", global_notifyhold)
 		M_BOOL("alwaysauthreject", global_alwaysauthreject)
@@ -17530,6 +17538,9 @@
 
 	if (!p) 
 		return -1;
+	/* Disable early RTP bridge  */
+	if (chan->_state != AST_STATE_UP && !global_directrtpsetup)     /* We are in early state */
+		return 0;
 	sip_pvt_lock(p);
 	if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) {
 		/* If we're destroyed, don't bother */



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