[asterisk-users] Help with IP Routing

[Nivin Kumar]

Is there a tool that will allow me to automatically change sip headers in realtime?

--- On Wed, 26/5/10, Motiejus Jakštys <desired.mta at gmail.com> wrote:


From: Motiejus Jakštys <desired.mta at gmail.com>
Subject: Re: [asterisk-users] Help with IP Routing
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Date: Wednesday, 26 May, 2010, 1:17 PM


Assume previous IP is LAN. Forwarding public IP ports to LAN is
straighforward. However, with SIP headers you will (don't know H323)
have to modify outgoing SIP headers: replace LAN ip with WAN ip.
For callers you have to substitute RTP destination IP
For callees you have to substitute RTP source IP.

I`m afraid you will have to check more details here:
http://www.ietf.org/rfc/rfc3261.txt
Maybe client sends server it's own IP address?

However, dumb header substitution + port range forwarding should work
in all cases for SIP.

On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar <nivinkumar24 at yahoo.in> wrote:
>
> Hello,
>
> I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions?
>
> Thanks,
> Nivin
>
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