[Freeswitch-users] rtp natting

[Kristian Kielhofner]

Keep doing what you are doing with OpenSIPS.  Add FreeSWITCH for
things it's good at:

- SBC functionality
- Hosted voice apps (like conferencing)

FreeSWITCH is good at those other things too but the SERs are still
king of scale.

On Fri, Aug 26, 2011 at 10:08 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi,
>
>
>
> I am new to freeswitch and wish to bolt it into our existing network to
> provide conferencing services.
>
>
>
> Currently we just use an opensips/rtpproxy configuration for simple proxy
> switching.
>
>
>
> However looking at the features of freeswitch it seems to me that I could
> replace my entire setup with just the freeswitch platform. Would that be a
> fair comment?
>
>
>
> If so then there a number of things that I am struggling to find clearly
> defined in the documentation.
>
>
>
> 1.       Currently our rtp proxy scenario issues a port to come back on but
> waits for incoming rtp packets to determine the exact port to return rtp on
> when the caller is behind a nat. Does freeswitch do this, does it need to be
> configured to work that way? Freeswitch would be on a public address.
>
> 2.       Currently opensips stores all user info in a mysql database and
> with 500000+ users is easy to manipulate. How do you deal with such a large
> user database in a pure xml environment such as freeswitch?
>
>
>
> Many thanks for any help you can offer.
>
>
>
> Regards
>
>
>
> Chris
>
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>



-- 
Kristian Kielhofner