diff options
| author | Olle Johansson <oej@edvina.net> | 2007-02-02 00:26:25 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2007-02-02 00:26:25 +0000 |
| commit | cfe66e6b26931c9cd92f2f4f1d36694b1b8baad6 (patch) | |
| tree | bfdbfed71e156ca94b0c7eeaad4514a52cd504c8 /channels | |
| parent | 44a9af35761c45010dc320c4786d683bdadaefb3 (diff) | |
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
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 10 |
1 files changed, 10 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 4767ac6a1..0cf2215bf 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 */ |