SIP-NAT issues (was: RE: [Asterisk-Users] Free World Dialup behind NAT)

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*
sip.conf for this is as follows:

[fwd1]
reinvite=no
canreinvite=no
nat=yes
type=friend
secret=dunk13
username=33537
host=fwd.pulver.com
;host=192.246.69.247
context=inbound

If i have 'canreinvite' omitted or yes, call setup doesn't work at all.


Regards,

Jamie Carl
Email:	me at jazz-inc.net
PH:		+61-414-365-466

-----Original Message-----
From: William Walsh [mailto:william at wxw.org]
Sent: Sunday, 25 May 2003 4:59 PM
To: asterisk-users at lists.digium.com
Subject: Re: SIP-NAT issues (was: RE: [Asterisk-Users] Free World Dialup
behind NAT)


*This message was transferred with a trial version of CommuniGate(tm)
Pro*

In your sip client context in sip.conf, do you have:
canreinvite=no

If not, add it, and try again.


On Sat, 2003-05-24 at 22:28, Jamie Carl wrote:
> *This message was transferred with a trial version of CommuniGate(tm)
Pro*
> I've been trying this too.  I've had a look at the SIP packets and the
> SDP section gives the internal IP address for RTP destination.
> 
> INVITE sip:10001 at fwd.pulver.com SIP/2.0
> Via: SIP/2.0/UDP 10.50.1.2:5060;branch=z9hG4bK47758f0a
> From: "asterisk" <sip:asterisk at 10.50.1.2>;tag=as6f148dbb
> To: <sip:10001 at fwd.pulver.com>
> Contact: <sip:asterisk at 10.50.1.2>
> Call-ID: 671df30d248a69495aaa784f64aa6fc3 at 10.50.1.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Content-Type: application/sdp
> Content-Length: 183
> 
> v=0
> o=root 10741 10741 IN IP4 10.50.1.2
> s=session                 ^^^^^^^^^
> c=IN IP4 10.50.1.2
>          ^^^^^^^^^^  <--  Here's the problem
> t=0 0
> m=audio 11296 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> 
> So call setup works, but the remote endpoint is sending the RTP stream
> to the wrong address.  Is there any way around this??
> 
> I have "nat=yes" in my sip.conf file, but there seems to be NO public
> information in this header at all.  So i'm thinking, what does
"nat=yes"
> actually do?
> 
> 
> Regards,
> 
> Jamie Carl
> Email:	me at jazz-inc.net
> PH:		+61-414-365-466
> 
> -----Original Message-----
> From: Shaun Ewing [mailto:shaun at ewing.dropbear.id.au]
> Sent: Sunday, 25 May 2003 2:07 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Free World Dialup behind NAT
> 
> 
> *This message was transferred with a trial version of CommuniGate(tm)
> Pro*
> 
> ----- Original Message -----
> From: "Oliver Brandt" <oliver_mlisten at gmx.de>
> 
> <SNIP>
> 
> > Anyway, I set up an acount and as long as my * box dials into the
> > internet itself it works fine. But as soon as I try to connect from
> > behind the NAT I can't here the other person (he can here me
though).
> >
> > My setup:
> > ATA -> * -> NAT -> Freeworld -> other person
> 
> I had the same problem.
> 
> I tried everything I could think of (forwarding ports, etc) with no
> luck.
> 
> It seemed (by looking at sip debug) that Asterisk was including its
> internal
> IP address in the outgoing headers. My guess is that the udp reply
> packets
> are being sent to that internal address - but as the person isn't on
> your
> network they're going into a black hole.
> 
> The only way I could solve it was by using one of my public IP
addresses
> (my
> ISP gives up to 4).
> 
> Previously my config was similar to yours, but now it is:
> 
> Softphone -> * -> FWD -> other person
>                  -> * -> LAN (other phones).
> 
> Once I had the * box on the public Internet and not behind NAT it
worked
> perfectly.
> 
> It would be interesting to see if there is a way to get around this
> without
> having an IP - I'm not too keen on having this box on the public
> Internet;
> I'd feel much more secure with it behind the firewall.
> 
> --Shaun
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> 
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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-- 
William Walsh <william at wxw.org>
Jabber: william at wxw.biz


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