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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 4767ac6..0cf2215 100644 (file)
--- 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 8307399..df22910 100644 (file)
--- 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