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authorOlle Johansson <oej@edvina.net>2006-06-09 09:47:44 +0000
committerOlle Johansson <oej@edvina.net>2006-06-09 09:47:44 +0000
commit9f5aa13142006f51f9ac5bf98ca98ebc97d69e82 (patch)
tree281dc522d9e283fb94496ba3224c78aab4ac5a4c /include/asterisk/rtp.h
parentce10b34ca68c02cd0ed647ef5cea964bbe79232e (diff)
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
Diffstat (limited to 'include/asterisk/rtp.h')
-rw-r--r--include/asterisk/rtp.h4
1 files changed, 3 insertions, 1 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index b7be53a1c..3bd105169 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);