[asterisk-users] NAT

[Stefan van der Eijk]

On 7/9/07, Noah Miller <noahisaacmiller at gmail.com > wrote:
>
> Hi Stefan -
>
> > What I want to accomplish:
> > - calls within the LAN are re-invited (RTP goes from endpoint to
> endpoint)
> > - asterisk detects when a call is going beyond the local LAN (over the
> NAT),
> > and then stays in the middle.
> >
> > I'm wondering if this is hard to do and how I'm supposed to configure
> this.
>
> I don't really know how hard it would be to do what you describe, but
> if you're interested in getting the results you want with a minimum of
> effort, just keep asterisk in the media path all the time.  Set
> canreinvite=no, and your calls should work consistently whether they
> stay inside the NAT or go outside.


This is what I ended up doing. Until I ran into issues again with outgoing
calls. Current setup = asterisk 1.4.11, installed on a host connected to the
internet (internet route able IP-address) and my internal network (
192.168.254.254). SIP phones are on the internal network, STUN and such
hasn't been configured.

SIP.conf:
externhost = <external hostname --> ddns.org>
canreinvite = no
localnet = 192.168.254.0/24
; nat = option is not set

Outgoing call to our sip provider ends up being setup like this:

outbound RTP stream:
SIP phone (192.168.254.104) --> asterisk internal (192.168.254.254)
asterisk external (internet IP) --> asterisk external (internet IP) (!!!)

inbound RTP stream:
SIP provider (internet IP) --> asterisk external (internet IP)
asterisk internal (192.168.254.254) --> SIP phone (192.168.254.104)

I have no idea why asterisk is trying to send the outbound RTP stream to
itself. Removing the externhost and localnet settings doesn't help either.
Neither does setting "nat = yes", even in the example below.




SIP.conf:
externhost = <external hostname --> ddns.org>
canreinvite = nonat
localnet = 192.168.254.0/24
; nat = option is not set.

Outgoing call to our sip provider ends up being setup like this:

outbound RTP stream:
SIP phone (192.168.254.104) --> asterisk internal (192.168.254.254)
asterisk external (internet IP) --> SIP provider (internet IP)

inbound RTP stream:
SIP provider (internet IP) --> asterisk external (internet IP)
asterisk internal (192.168.254.254) --> SIP phone (192.168.254.104)

The inbound RTP stream goes well for +/- 1 second, then the SIP provider
responds to a re-invite sent by my asterisk box to send the trafic to
192.168.254.104 (the SIP phone on my internal network).

outbound RTP stream:
SIP phone (192.168.254.104) --> asterisk internal (192.168.254.254)
asterisk external (internet IP) --> SIP provider (internet IP)

inbound RTP stream:
SIP provider (internet IP) --> SIP phone (192.168.254.104)

I don't understand the logic of Asterisk sending the re-invite for inbound
RTP stream. I would be more logical if Asterisk would send an invite for the
outbound RTP stream:

outbound RTP stream:
SIP phone (192.168.254.104) --> SIP provider (internet IP)

inbound RTP stream:
SIP provider (internet IP) --> asterisk external (internet IP)
asterisk internal IP (192.168.254.254) --> SIP phone (192.168.254.104)

Does the logic have anything to do with in which order the interfaces are
defined on the box? In my case, ETH0 = 192.168.254.254, ETH1 = internet IP.

I can't find any configuration examples of my kind of setup, where a
dual-homed host running asterisk has one NIC on the Internet and one on the
internal (RFC1918 space) network. All examples I've bumped into have either
the asterisk box behind a NAT router (i.e. it only has a RFC1918 IP-address)
or the asterisk box is on a real IP.

with kind regards,

Stefan
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