[asterisk-users] one way RTP using NAT

[srgqwerty]

I am not an Asterisk expert.
But in had a similar problem (using a softphone instead of your Cisco).

I corrected this issue modifying one parameter in the SOFTPHONE (not the 
asterisk).
After changing this parameter all starts to be OK (RTP are send to the public 
IP instead of the private one).

The softphone was "twinkle" for linux and the parameter is at "edit/user 
profile/nat" and select "use static configured ip address inside SIP 
messages".

Before that change the SIP negotiation was OK but no RTP packets was arriving 
at the phone (because the asterisk was sending those packets to the private 
IP of the phone). You can see this with a sniffer or using "rtp debug" in the 
asterisk console.

Also note that I think that this problem can be solved using "inteligent" NAT 
devices.
This problem is similar to the FTP data connection in what the client tells 
the server "I am listening for your connection in <IP> at <PORT>", of course 
the <IP> is the REAL IP of the client.
Today, all NAT devices are capable of changing this message at layer 7 to be 
like:
"I am listening for your connection at <NAT_IP> at <NAT_PORT>".

Hope this helps.

Regards

On Friday 02 November 2007 21:01, Hector Quiroz wrote:
> Hi,
> I'm having a problem with my asterisk, trying to connect to a CISCO 2840
> IOS12.x ASterisk is behind firewall NATing, when it do the handshaking for
> RTP, it sends his internal IP instead of sending the external one. How can
> I tell the asterisk box, to modify that and send the external IP? I tryied
> with Sip.conf's externip=xxxx and localnet=xxxx, nat=yes
> Nothing seems to change the contact field on SIP protocol going out
> from the asterisk box..
>
> thanks.
> Hector.
>
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