Derek,
I'm not an expert in these area's hence the offer to play, but in answer
to your questions to the best of my ability -
1. I don't see any reason the outbound proxy cant be in the public
domain although this is where the NAT issues start kicking in
(especially if you want incoming calls), depending on the number of
clients behind the firewall you would have to do lots of port mapping
etc. on the router/firewall, could be done but would be painful.
2. Never played with a STUN server, sorry just another point to break in
the chain?
_______________________________________________
Ray
_______________________________________________
-----Original Message-----
From: Derek Conniffe [mailto:[EMAIL PROTECTED]
Sent: 13 September 2005 17:50
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Nat & Sip & Pain
Hi Ray,
It would be great to find a solution which doesn't need modification of
the firewall setup (like if it was a customers firewall rather than your
own).
There is two things I'm wondering about: -
1) Can a "Outbound SIP Proxy" be a server out on the Internet (i.e. not
in the local network this side of the NAT) and does that provide a way
to make the SIP via NAT work? *
2) Is STUN a workable solution. There is no problem running a STUN
server but can the far side of the STUN connection (Internet) talk with
Asterisk and is this a way to make the SIP via NAT work? **
* I would have thought that an "Outbound Proxy" would need to be inside
on the local network (a bastion host rather like a squid server for
HTTP) but then I read the FWD documentation about setting the Outbound
Proxy for a budgetone to make it work with NAT and their server - the
Outbound Proxy they specified was out there on the Internet.
** I've read that Asterisk doesn't currently have STUN support but I'm
not sure what that means exactly: I'm not sure if that means "Asterisk
doesn't have an STUN server built-in" or if it means "Asterisk is not
compatible with an STUN server".
Thanks,
Derek
razza wrote:
Derek,
You said -
Needless to say when I don't have any NAT settings on the SIP phone I
don't get any registration with the * server (this confuses me too -
I'm
not sure why I only get registration when I set the * server to be the
outbound proxy? Maybe its because the SIP phone sends its local IP in
the RTP packets?).
SIP is not NAT friendly (unlike IAX) and yes your device will try to
send its local IP (in SIP packets), unless in the case of a budgetone
phone you set the 'Use NAT IP' to your external IP addr. You will also
have to NAT the public ip for the SIP port (5060?) and RTP ports
(whatever) to your phones private IP.
Must admit not tried it myself, but happy to jointly experiment if you
like?
_______________________________________________
Ray
_______________________________________________
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Conniffe
Sent: 13 September 2005 12:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Nat & Sip & Pain
Hi everyone,
I decided to have a look at SIP & NAT again and I've been at it for a
[quite a] few hours but typically nothing is working for me. Actually
I'm not sure if SIP and NAT can ever work but some emails on this list
do suggest that someone has got it working, once, maybe.
I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports
"Outbound Proxy", "STUN" and "Fake WAN Address on SIP and RTP". I'm
using Netfilter (IPTables) on Linux as the Firewall at NAT gateway to
the Internet.
I'm lacking knowledge in UDP, RTP and SIP - which doesn't help of
course.
In my experiments the only thing that seems to allow me to make a call
is to enter the [public Internet] IP address of my * server into the
"Outbound Proxy" setting in the SIP phone - then it registers and I can
make a call but no audio, either direction, is heard.
I would have thought that the "Outbound Proxy" should be inside the NAT
gateway but then I read the settings for a Budgetone BEHIND nat on the
FWD webpage
(http://www.freeworlddialup.com/support/configuration_guide/configure_y
o
ur_fwd_certified_phone/grandstream_budgetone/outbound_proxy)
where they suggest that the Outbound Proxy should be an external
Internet public proxy server ?
Then I was reading about STUN and what a nice sounding solution it is -
so I downloaded and installed the Vivida STUN server - compilation &
installation was nice and easy and I set the STUN primary IP address &
port into the SIP phones STUN servers settings. I could see that the
SIP phone communicated with the STUN server (lots of stuff about
mapping
between my local NAT gateway's public IP address and the secondary IP
address of the STUN server)... but no registration or [apparent]
communication with the * server.
I didn't try to do anything with the "Fake WAN address.." settings or
try to redirect incoming UDP ports from the firewall to the SIP phone
because I'm trying to see if its possible to setup a deploy-anywhere
SIP
phone solution.
Needless to say when I don't have any NAT settings on the SIP phone I
don't get any registration with the * server (this confuses me too -
I'm
not sure why I only get registration when I set the * server to be the
outbound proxy? Maybe its because the SIP phone sends its local IP in
the RTP packets?).
Does anyone know how to get NAT & SIP working where the SIP phone is
behind a NAT server talking to a publicly accessible * server?
Thanks for any help!
When I run FWD's "netcheck" on my local PC (also behind the NAT) I get:
Internet Connection: Connected, Direct/NAT: Using NAT, NAT type: Port
Restricted Nat, NAT UPnP enabled: No, Local IP Address: 192.168.5.10,
WAN IP Address: XXX.XXX.XXX.XXX (public IP address), Port 5060:
Blocked,
port 5082: Blocked.
[Maybe] useful Links that I've found on my Nat & SIP travels:-
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
-------------------------------------------------------------
Here VOIP INFO claim that "Asterisk as a SIP server outside nat,
clients
on the inside connecting to Asterisk" is "solved" with "with nat
<tiki-index.php?page=Asterisk+sip+nat>=yes and qualify
<tiki-index.php?page=Asterisk+sip+qualify>=xxx in sip.conf
<tiki-index.php?page=Asterisk+config+sip.conf> for the client in most
cases. Some clients (X-lite) assist themselves by using STUN
<tiki-index.php?page=STUN> and sending UDP keep-alive packets. Qualify
<tiki-index.php?page=Asterisk+sip+qualify> sends keep-alive packets
from
Asterisk to the client on the inside." - however I can't get it to work
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asteri
s
k.html
-----------------------------------------------------------------------
-
-----------
Here there is some detail about the NAT= option in sip.conf and
firewall
NAT types plus some understandable diagrams of why SIP & NAT is so much
bother.
http://www.voip-info.org/wiki-STUN
--------------------------------------
The VOIP INFO page about STUN - I don't think I learned much here -
except the link to the Vovida STUN server software
Asterisk Users - Email from [EMAIL PROTECTED] - 02/July/2005 23:49
--------------------------------------------------------------------
Thierry claims that you need to put special MASQUERADE POSTROUTING
rules
into iptables to make it NAT UDP properly - tried it but didn't work
for me
Asterisk Users - Email from [EMAIL PROTECTED] - 16/Aug/2005 10:29
-----------------------------------------------------------------------
-
Kamran Ahmad sounds like someone who [might have] had SIP & NAT working
- until it wasn't working....
BTW My Current SIP sip.conf entry that I'm using for testing (which
doesn't work of course!): -
[0035314401789]
context=PublicSip
type=friend
port=5060
username=0035314401789
password=XXXXXXXX
callerId=0035314401789
nat=route ; assume a NAT connection (note: route
doesn't seem to make any difference compared to "yes")
qualify=yes ; keep-alive packets to keep NAT SIP
open
insecure=yes ; insecure and auth don't seem to
make things work any better/worse!
auth=plaintext ;
host=dynamic ; and with a dynamic IP address
canreinvite=no ; always keep asterisk in the media
path
;dtmfmode=info ; could be inband ?
dtmfmode=rfc2833 ; could be inband ? but doesn't matter
-
still NAT & SIP isn't working
[EMAIL PROTECTED]
disallow=all
;allow=ilbc
;allow=ulaw
allow=g729
;allow=ulaw
;allow=all
