So is the answer ultimately that it cannot handle any packet loss or a tiny bit in order to not affect user to user calls over the Internet?
On Mon, 22 Mar 2010 20:24:12 -0400, Tony Graziano wrote: > On Mon, Mar 22, 2010 at 8:16 PM, Josh Patten <[email protected]> > wrote: > >> Media relay is used for NAT traversal purposes. I am pretty sure if you >> are using sipX for nat traversal then the media path is phone<--->sipX<--- >> >phone >> >> > Ding!Ding!Ding! Correct answer. > >> This is necessary to overcome firewall issues methinks. >> >> > "IF" you are using an independent SBC for trunking and/remote users you are > not enabling either of the sipXbridge or Media Relay functions. In either > case it's a matter of how the media is setup, and there are numerous ways > this can happen, depending on the network schema. > > > 1. VPN with all traffic routing via the VPN to call another internal user > (whether using the VPN or sitting on the native sip Lan, is not using media > relay or SBC of any kind. > 2. ANY remote user using remote NAT traversal within sipx is anchoring > media via sipXrelay, no matter where the call is going (AA, VM, another > internal user, PSTN. > 3. sipXbridge is only used as a SBC for routing siptrunk calls (PSTN). Any > remote user anchors their media TO sipx then OUT to PSTN via sipXridge, > using twice the bandwidth. > > > I can only draw stick people animations, and they are not pretty, so I'll > leave it at that. > > >> Todd Hodgen wrote: > My vote is its UA to UA since the Bridge is not >> involved. -----Original Message----- From: sipx-users- >> [email protected] [mailto:sipx-users- >> [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: sipx- >> [email protected] [mailto:sipx-users- >> [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/ >> >> >> _______________________________________________ >> 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/
