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authorJoshua Colp <jcolp@digium.com>2006-11-30 21:22:01 +0000
committerJoshua Colp <jcolp@digium.com>2006-11-30 21:22:01 +0000
commit869101028b619423f7c4d78ee96d3b669efc8cb4 (patch)
tree3228da5ddf405a44a55083a5cc1979f85561a5c9 /include/asterisk
parent813f8d26fec6286542e3d202cc4f1bb6c9fc5a90 (diff)
Merged revisions 48168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include/asterisk')
-rw-r--r--include/asterisk/rtp.h3
1 files changed, 3 insertions, 0 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 03f204453..aab598a31 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -186,6 +186,9 @@ void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
/*! \brief Compensate for devices that send RFC2833 packets all at once */
void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
+/*! \brief Enable STUN capability */
+void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
+
int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
int ast_rtp_proto_register(struct ast_rtp_protocol *proto);