Re: [asterisk-users] NAT

On 7/9/07, Noah Miller <noahisaacmiller@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