Gabriel Ortiz Lour wrote: > Hi all, > > Suposing that 2 SIP phone register at a remote (internet) asterisk, > what is the best way, if any, to make the RTP traffic go phone to phone, > whithout using the internet conection (asterisk)? > > Thanks, > Gabriel >
By default, Asterisk will attempt to offload the media from the server so that it may flow directly between the phones. There are several factors which may prevent this, though. For instance, if Asterisk is recording the call or needs to listen for DTMF in order to activate a specific feature, then Asterisk has to have the RTP flow through it. Mark Michelson _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
