[asterisk-users] Sending RTP media to a different server than SIP Signaling

[bruce bruce]

There you go. This confirms that SIP signaling determines where the calls
should go. I would take their word with a grain of salt specially with their
whole support center our of India. No disrespect, but it is bad service
overall.

-Bruce

On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp <jcolp at digium.com> wrote:

> ----- "Tarek Sawah" <tareksawah at hotmail.com> wrote:
>
> > we started with them two days ago .. and we are facing plenty of False
> > Answer cases on several destinations although ppl said they have a
> > policy against FAS..
> > anyway i don't know i will be looking into another method to send the
> > RTP to another server,
>
> The IP address (and port) of where to send audio is negotiated when
> the call is setup. You can't change it or specify an IP address to use.
> Even if you did change the IP address you would be sending it to the port
> associated with the session on the other media gateway. That would just
> not work.
>
> --
> Joshua Colp
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
> --
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