diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-24 20:29:41 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-24 20:29:41 +0000 |
| commit | 10edb20a8e659e7a8468ec032aa5042fd70b6d86 (patch) | |
| tree | 1affbf2f9fe7a7ec24e319146b7094431b6707e1 /main/rtp.c | |
| parent | 2c162efa7e1633695c3a96027f3aefc7e00910c3 (diff) | |
Doxygen additions, corrections
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56665 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 10 |
1 files changed, 6 insertions, 4 deletions
diff --git a/main/rtp.c b/main/rtp.c index afb5c256a..728eaa849 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -3277,10 +3277,6 @@ static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast return res; } -/*! \brief Bridge calls. If possible and allowed, initiate - re-invite so the peers exchange media directly outside - of Asterisk. -*/ /*! \page AstRTPbridge The Asterisk RTP bridge The RTP bridge is called from the channel drivers that are using the RTP subsystem in Asterisk - like SIP, H.323 and Jingle/Google Talk. @@ -3306,6 +3302,12 @@ static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast - ast_rtp_bridge() - ast_channel_early_bridge() - ast_channel_bridge() + - rtp.c + - rtp.h +*/ +/*! \brief Bridge calls. If possible and allowed, initiate + re-invite so the peers exchange media directly outside + of Asterisk. */ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms) { |