diff options
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 29 |
1 files changed, 28 insertions, 1 deletions
diff --git a/main/rtp.c b/main/rtp.c index c629da006..7163f1601 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -3142,7 +3142,34 @@ static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast /*! \brief Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside - of Asterisk. */ + 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. + + This bridge aims to offload the Asterisk server by setting up + the media stream directly between the endpoints, keeping the + signalling in Asterisk. + + It checks with the channel driver, using a callback function, if + there are possibilities for a remote bridge. + + If this fails, the bridge hands off to the core bridge. Reasons + can be NAT support needed, DTMF features in audio needed by + the PBX for transfers or spying/monitoring on channels. + + If transcoding is needed - we can't do a remote bridge. + If only NAT support is needed, we're using Asterisk in + RTP proxy mode with the p2p RTP bridge, basically + forwarding incoming audio packets to the outbound + stream on a network level. + + References: + - ast_rtp_bridge() + - ast_channel_early_bridge() + - ast_channel_bridge() +*/ 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) { struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */ |