[asterisk-users] Asterisk and Client NAT

[G B]

Hi Gordon,

I did everything that you suggested, however, the symptoms remain.

I set the rtp.conf to use ports 10000 to 20000

I assured that my router was forwarding these ports. However, the Media Description Section of the SIP/SD packet (captured with ethereal) reads:

Media Description, name and address (m): audio 50486 RTP/AVP 0 8 101

50486 is the destination port of all RTP packets sent from the client. These are filtered out by my server NAT's firewall. It seems that Asterisk is not using rtp.conf

I did some searching and found the following link. This is right around the time that I downloaded. Could this be the trouble?

http://lists.digium.com/pipermail/asterisk-bugs/2007-July/001213.html

> Date: Sun, 19 Aug 2007 11:08:57 +0100
> From: gordon+asterisk at drogon.net
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Asterisk and Client NAT
> 
> On Sun, 19 Aug 2007, G B wrote:
> 
> >
> > Hi,
> >
> >
> > I realize that this is amongst the worst configurations, but I have been 
> > made to believe that it can work... eventually. However, currently SIP 
> > call set up seems to go fine, but no media is transferred in either 
> > direction. For example, the following is output on the asterisk CLI 
> > despite no voice being heard. -- Executing [101 at john:1] 
> > Playback('SIP/john-081da978', 'hello-world') in new stack
> 
> *sigh* The old NAT & SIP issue - again... )-:
> 
> There is a lot of the VoIP WiKi on it. Eg:
>    http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
> 
> However, assuming the asterisk and client are behind different NAT 
> firewalls, do this:
> 
> 1. Tell the client to use a stun server and don't fiddle with the client's 
> firewall (other than to make sure it's not actually firewalling 5060 and 
> 10000-20000)
> 
> If you're stuck for a stun server, use stun1.drogon.net:3478
> 
> 2. Port forward 5060-5069 and 10000-20000 on the firewall that fronts the 
> asterisk box to the asterisk box.
> 
> 3. Tell asterisk it's behind a NAT firewall.
> 
> > 1. sip.conf
> > [global]
> > nat=yes
> > canreinvite=no
> 
> This isn't enough. You also need to tell it the IP address of the external 
> firewall, and your local network address.
> 
>    nat=yes
>    localnet=192.168.2.0/24
>    externip=1.2.3.4
> 
> Where 1.2.3.4 is the external IP address - the one the client is pointing 
> to. This needs to be a static IP address (or at least not change for the 
> duration of your use) the client can be behind a dynamic IP address.
> 
> you might need a bit more in the client definition - eg:
> 
> [100]
> context=internal
> type=friend
> secret=very
> qualify=yes
> nat=yes
> host=dynamic
> canreinvite=no
> dtmfmode=rfc2833
> mailbox=100
> callerid=Joe Bloggs <100>
> callgroup=1
> pickupgroup=1
> subscribecontext=BLF
> 
> And that's it.
> 
> Gordon
> 
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