From 9f5aa13142006f51f9ac5bf98ca98ebc97d69e82 Mon Sep 17 00:00:00 2001
From: Olle Johansson <oej@edvina.net>
Date: Fri, 9 Jun 2006 09:47:44 +0000
Subject: [PATCH] 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
---
apps/app_dial.c | 12 ++++++------
include/asterisk/rtp.h | 4 +++-
rtp.c | 2 +-
3 files changed, 10 insertions(+), 8 deletions(-)
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 56855337b7..5361bdc6a9 100644
--- 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 b7be53a1ca..3bd1051693 100644
--- 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 8291374920..642c8954e9 100644
--- 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 */
--
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