[OpenSIPS-Users] R: R: opensips to balance 2 freeswitch

[Alessandro Illiano]

Hi,
Ok I'm going to try, a simple question: 
Incoming call works fine with opensips+rtpproxy in bridged mode.
But what if a second call is generated directly from freeswitch and bridged
to the frist one?
The second call flow bypass opensips+rtpproxy but goes directly to the
destination (at the moment using opensips server as gateway using
net.ipv4.ip_forward = 1 kernel parameter).
It's right?

Regards,
Alessandro

 


-----Messaggio originale-----
Da: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] Per conto di Ovidiu Sas
Inviato: giovedì 20 gennaio 2011 16:55
A: OpenSIPS users mailling list
Oggetto: Re: [OpenSIPS-Users] R: opensips to balance 2 freeswitch

Just enable the nathelper module along with rtpproxy configured in
bridge mode and it will work fine.

Regards,
Ovidiu Sas

On Thu, Jan 20, 2011 at 10:45 AM, Alessandro Illiano
<alessandro.illiano at neexa.it> wrote:
> 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: users-bounces at lists.opensips.org
> [mailto:users-bounces at lists.opensips.org] 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
>>
>>
>>
>>
>>
>>
>> _______________________________________________
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>> Users at lists.opensips.org
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>>
>>
>
>
> --
> 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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