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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-11-30 21:18:24 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-11-30 21:18:24 +0000
commit260b85dbbc8de8357490cc0e26eb73f09f210943 (patch)
tree50e05458979ea7ce5639c4ef524257e3ea958b34 /include
parent74ad2ee9b28a6c2ae1560ab25fd344f24fa2fb54 (diff)
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48168 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
-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 c332159ec..d7738b345 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);