From cfe66e6b26931c9cd92f2f4f1d36694b1b8baad6 Mon Sep 17 00:00:00 2001
From: Olle Johansson <oej@edvina.net>
Date: Fri, 2 Feb 2007 00:26:25 +0000
Subject: [PATCH] Patch based on this patch with small changes for trunk...
Merged revisions 53109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines
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/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 10 ++++++++++
configs/sip.conf.sample | 6 ++++++
2 files changed, 16 insertions(+)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 4767ac6a16..0cf2215bf6 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -526,6 +526,7 @@ 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 */
@@ -10478,6 +10479,7 @@ 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)"));
@@ -16337,6 +16339,7 @@ static int reload_config(enum channelreloadreason reason)
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);
@@ -16463,6 +16466,8 @@ 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")) {
@@ -17013,6 +17018,11 @@ 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;
+
sip_pvt_lock(p);
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 83073996d6..df22910a9a 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -333,6 +333,12 @@ 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
--
GitLab