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| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-11-30 21:22:01 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-11-30 21:22:01 +0000 |
| commit | a9383ac9273d884e8bde5319ff8c77d65bd966f9 (patch) | |
| tree | 3228da5ddf405a44a55083a5cc1979f85561a5c9 /include/asterisk/rtp.h | |
| parent | b631ec45c7acd9827e749b787b4a4eb11e83ad69 (diff) | |
Merged revisions 48168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48169 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include/asterisk/rtp.h')
| -rw-r--r-- | include/asterisk/rtp.h | 3 |
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); |