From: Olle Johansson <oej@edvina.net>
Date: Fri, 2 Feb 2007 00:24:03 +0000 (+0000)
Subject: Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but... 
X-Git-Tag: 1.4.1~148
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=90a4b844a97ca87576a37942d5f8f17b72e45822;p=thirdparty%2Fasterisk.git

Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 22953bbcf0..2f2d8ba4ff 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -523,6 +523,7 @@ static int default_maxcallbitrate;	/*!< Maximum bitrate for call */
 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 */
@@ -10241,6 +10242,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
 	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)"));
@@ -16080,6 +16082,7 @@ static int reload_config(enum channelreloadreason reason)
 	expiry = DEFAULT_EXPIRY;
 	global_notifyringing = DEFAULT_NOTIFYRINGING;
 	global_limitonpeers = FALSE;
+	global_directrtpsetup = FALSE;		/* Experimental feature, disabled by default */
 	global_notifyhold = FALSE;
 	global_alwaysauthreject = 0;
 	global_allowsubscribe = FALSE;
@@ -16206,6 +16209,8 @@ static int reload_config(enum channelreloadreason reason)
 			ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
 		} else if (!strcasecmp(v->name, "limitonpeers")) {
 			global_limitonpeers = ast_true(v->value);
+		} else if (!strcasecmp(v->name, "directrtpsetup")) {
+			global_directrtpsetup = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "notifyringing")) {
 			global_notifyringing = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "notifyhold")) {
@@ -16757,6 +16762,11 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 	p = chan->tech_pvt;
 	if (!p) 
 		return -1;
+
+	/* Disable early RTP bridge  */
+	if (chan->_state != AST_STATE_UP && !global_directrtpsetup) 	/* We are in early state */
+		return 0;
+
 	ast_mutex_lock(&p->lock);
 	if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) {
 		/* If we're destroyed, don't bother */
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 2948eebabd..293eafa636 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -307,6 +307,12 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; call directly between the endpoints instead of sending
 				; a re-INVITE).
 
+;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 
+				; RTP payloads and fmtp headers in the 200 OK that does not match the
+				; callers INVITE.
+
 ;canreinvite=nonat		; An additional option is to allow media path redirection
 				; (reinvite) but only when the peer where the media is being
 				; sent is known to not be behind a NAT (as the RTP core can