[asterisk-users] Asterisk and Client NAT

[G B]

Thank you very much for your prompt replies. Perhaps I will consider moving to a 1.2 version of Asterisk. 

> Date: Sun, 19 Aug 2007 12:08:36 +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 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
> 
> I know what when I do that on my systems, it "just works". Even with 
> xlite.
> 
> I've never fiddled with rtp.conf. Mine is as it came with the default 
> installation.
> 
>    rtpstart=10000
>    rtpend=20000
> 
> However, I'm running asterisk version 1.2.X, so there might be some other 
> issues with 1.4.
> 
> This is the scenario that 99% of my installations work under for people 
> with phones not on the office LAN, and so-far so good (for me!)
> 
> Gordon
> 
> >
> >> 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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