Rename ast_rtp_early_media to ast_rtp_early_bridge to avoid confusion.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33174 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 56855337b7d313c5f6a500e970c2527a046c2295..5361bdc6a9122bf35c118bab83d72cba2fe172d5 100644 (file)
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -576,8 +576,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
                                                               OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
                                                               OPT_CALLEE_PARK | OPT_CALLER_PARK |
                                                               DIAL_NOFORWARDHTML);
-                                               /* Setup early media if appropriate */
-                                               ast_rtp_early_media(in, peer);
+                                               /* Setup RTP early bridge if appropriate */
+                                               ast_rtp_early_bridge(in, peer);
                                        }
                                        /* If call has been answered, then the eventual hangup is likely to be normal hangup */
                                        in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
@@ -606,7 +606,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
                                                ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name);
                                        /* Setup early media if appropriate */
                                        if (single)
-                                               ast_rtp_early_media(in, c);
+                                               ast_rtp_early_bridge(in, c);
                                        if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
                                                ast_indicate(in, AST_CONTROL_RINGING);
                                                (*sentringing)++;
@@ -617,7 +617,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
                                                ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", c->name, in->name);
                                        /* Setup early media if appropriate */
                                        if (single)
-                                               ast_rtp_early_media(in, c);
+                                               ast_rtp_early_bridge(in, c);
                                        if (!ast_test_flag(outgoing, OPT_RINGBACK))
                                                ast_indicate(in, AST_CONTROL_PROGRESS);
                                        break;
@@ -630,7 +630,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
                                        if (option_verbose > 2)
                                                ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name);
                                        if (single)
-                                               ast_rtp_early_media(in, c);
+                                               ast_rtp_early_bridge(in, c);
                                        if (!ast_test_flag(outgoing, OPT_RINGBACK))
                                                ast_indicate(in, AST_CONTROL_PROCEEDING);
                                        break;
@@ -1608,7 +1608,7 @@ out:
                sentringing = 0;
                ast_indicate(chan, -1);
        }
-       ast_rtp_early_media(chan, NULL);
+       ast_rtp_early_bridge(chan, NULL);
        hanguptree(outgoing, NULL);
        pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
        if (option_debug)

diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index b7be53a1cad22019756a9672bb6cd4ff49b1a29b..3bd105169349e6c070613279fee6bc2594ff8cec 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -182,7 +182,9 @@ void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
 
 int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
 
-int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src);
+/*! \brief If possible, create an early bridge directly between the devices without
+           having to send a re-invite later */
+int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src);
 
 void ast_rtp_stop(struct ast_rtp *rtp);
 

diff --git a/rtp.c b/rtp.c
index 82913749200a8a89fe7747e2be83898c15fdaacc..642c8954e9951bd755929117dbd8574837b41550 100644 (file)
--- a/rtp.c
+++ b/rtp.c
@@ -1274,7 +1274,7 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
        return cur;
 }
 
-int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src)
+int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
 {
        struct ast_rtp *destp, *srcp=NULL;              /* Audio RTP Channels */
        struct ast_rtp *vdestp, *vsrcp=NULL;            /* Video RTP channels */