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: [email protected]
> [mailto:[email protected] 
> <[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] 
> <[email protected]>] On Behalf [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.
>
>
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