On Wed, Mar 24, 2010 at 8:28 PM, [email protected] <[email protected]>wrote:
> 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/ > Any packet loss will result in a garble. A little bit more makes someone into darth vader. A little bit more means your call just got blasted into smithereens by the death star. There are other variations, moments of silence at ether end, etc. -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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