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Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now

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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
This commit is contained in:
Olle Johansson
2007-02-02 00:24:03 +00:00
parent b17706586f
commit 90a4b844a9
2 changed files with 16 additions and 0 deletions
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channels/chan_sip.c
+10
View File
@@ -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 */
configs/sip.conf.sample
+6
View File
@@ -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