Hi Bogdan,
Unfortunately i don't know... the problem is that we have many carriers
connected and now we accept traffic from their ip (only acl and fw rules to
prevent not authorized traffic).

May I solve configuring freeswitch servers enabling nat support?
But I don't know if this configuration can handle rtp flow correctly.

Maybe this is a wrong schema, so my question is if there are other solution
to manage many fs (or asterisk) server on a private lan with load balance.
(thinking lvs with direct routing but I think that sip header will contains
ip addr of the real server).

Regards,
Alessandro

-----Messaggio originale-----
Da: [email protected]
[mailto:[email protected]] Per conto di Bogdan-Andrei Iancu
Inviato: giovedì 20 gennaio 2011 13:01
A: OpenSIPS users mailling list
Oggetto: Re: [OpenSIPS-Users] opensips to balance 2 freeswitch

Hi Alessandro,

the problem is not the LB, but the presence of NAT between CARRIER and 
LB - any NAT presence must be explicitly handled in SIP....In your case 
is even more complicated as traffic comes from public, goes to private 
and again to public......

Does your carrier support "direction:active" or COMEDIA support for RTP ?

Regards,
Bogdan

Alessandro Illiano wrote:
> Hi all,
> i'm new to opensips. 
> I would like to know if it's possible to implement this scenario
>
> [CARRIER-A - internet]
> ---|---
> ---v--- 
> [opensips LB - internet ip + LAN IP 192.168.1.1] 
> ---|---
> ---v---
> [FS01 - ip 192.168.1.2] or [FS02 - ip 192.168.1.3] 
> ---|---
> ---v---
> [OTHER PBX - internet] or [local extension]
>
>
> Following load balancer mod tutorial, I've one way audio problem (routing
to
> other pbx). 
>
> Many thanks in advance,
>
> Alessandro
>
>
>
>
>
>
> _______________________________________________
> Users mailing list
> [email protected]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
OpenSIPS solutions and "know-how"


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