My vote is its UA to UA since the Bridge is not involved. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Picher, Michael Sent: Monday, March 22, 2010 4:21 PM To: [email protected]; sipx-users Subject: Re: [sipx-users] Packet loss Question
Two remote users (each behind NAT) through sipX is an interesting question as to whether the call is phone to phone or phone to pbx to phone. I believe the latter may be true. Ranga can answer that. If they were VPN'd in to the PBX the call would definitely be phone to phone. Mike -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Monday, March 22, 2010 6:52 PM To: sipx-users Subject: Re: [sipx-users] Packet loss Question On Mon, 22 Mar 2010 18:43:29 -0400, Picher, Michael wrote: > True internally on a network, but if you are talking about a sip trunk the > call will hairpin on the PBX and require 2x the bandwidth. I should have been more specific but yes, I mean two remote users, not two local users, over the Internet. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
