Re: [asterisk-users] Remote RTP
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
Re: [asterisk-users] Remote RTP
They will be in the same LAN, probably behind NAT. Being in the same LAN helps something? 2009/1/16 Jerry Jones > > On Jan 16, 2009, at 10:38 AM, 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)?They > > Allow reinvite? Assuming both are not behind NAT. > > > ___ > -- 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 > ___ -- 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
Re: [asterisk-users] Remote RTP
On Jan 16, 2009, at 10:38 AM, 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)? Allow reinvite? Assuming both are not behind NAT. ___ -- 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
Re: [asterisk-users] Remote RTP
canreinvite=yes. 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 > > > > > ___ > -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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
[asterisk-users] Remote RTP
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 ___ -- 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
