[OpenSIPS-Users] Load balancer setup

[Matt Broad]

Hi Kenny,

thanks for the reply.  For some reason this is only just showing in my
email so sorry for the delayed response.

Could you explain how you would do this is asterisk please? This may help
in me solving the issue :)



thanks
Matt

On 23 October 2014 21:40, Kenny Watson <KWatson at geniusppt.com> wrote:

>  Hi Matt,
>
> The SDP is being generated by freeswitch and you would need to make it
> think that when its sending to kamailio, that kamailio  is an external host
> so freeswitch uses its public IP address in the SDP  that it is then
> forwarded on directly to the carrier.
>
>  I've never used freeswitch but thats roughly what you'd do with asterisk.
>
> Thanks
> Kenny Watson
>
>
>   ------------------------------
> *From:* users-bounces at lists.opensips.org [users-bounces at lists.opensips.org]
> on behalf of matt [matt at supportedbusiness.com]
> *Sent:* 22 October 2014 08:42
> *To:* users at lists.opensips.org
> *Subject:* [OpenSIPS-Users] Load balancer setup
>
>   Hi,
>
>
>  I was looking for some guidance on using the load balancer in a NAT
> environment.
>
>  I have the following setup (the IP addresses are made up but should give
> an indication):
>
>  1 x opensips server with load balancer module - IP 192.168.0.1
>  2 x freeswitch servers - IP 192.168.0.2 & 192.168.0.3
>
>  All 3 servers have seperate external IP address routing to their
> internal IP via our firewall:
> 217.0.0.1 routed to 192.168.0.1 (Opensips)
> 217.0.0.2 routed to 192.168.0.2 (FS1)
>  217.0.0.3 routed to 192.168.0.3 (FS2)
>
>  I have the load_balancer table with the following details:
>
>  id,  | group_id, |                  dst_uri,            | resources,  |
> probe_mode, | description
> '1',  |      '1',     |  'sip:192.168.0.2:5080',  |   'pstn=10', |
>    '2',       |          'FS1'
> '2',  |      '1',     |  'sip:192.168.0.3:5080',  |   'vm=1',     |
>   '2',       |          'FS2'
>
>
>  The call flow is:
>
>  SIP Provider --> 217.0.0.1 Opensips --> 192.168.0.2/3
>
>  The issue is, that when the 200 ok response is sent to the SIP provider,
> the Freeswitch server's internal IP is being sent in the SDP connection
> information (c).  This causes the ACK response from the SIP Provider to
> fail to be sent correctly.
>
>  With the calls routed directly to the FS servers (removing opensips from
> the flow), the calls work fine.
>
>  Any help would be much appreciated :)
>
>
>  thanks
> Matt
>
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