Bridge calls. If possible and allowed, initiate re-invite so the peers exchange media directly outside of Asterisk.

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:

Asterisk is a trademark for Digium, inc.. | Edvina.net | Asterisk.org | This documentation was generated with Doxygen

Asterisk developer's documentation :: Codename Pineapple

The Asterisk RTP bridge