10 Jan
2003
10 Jan
'03
9:27 a.m.
What is this device? Where can I get one? What does it cost?
Thanks :-)
---greg
-----Original Message-----
From: serusers-admin(a)lists.iptel.org
[mailto:serusers-admin@lists.iptel.org] On Behalf Of Maxim Sobolev
Sent: Friday, January 10, 2003 8:15 AM
To: Jiri Kuthan
Cc: serusers(a)lists.iptel.org; kapitan(a)portaone.com
Subject: Re: [Serusers] Rewriting URI in the Contact field
Yes, I know - we have studied all those methods in details. Our
method of choice is symmetric signalling/symmetric media (aka COMEDIA)
due to the following reasons:
1. Things should work without modifying or reconfiguring existing
user's infrastructure (NATs) and should be compatible with all
widely-used NATs.
2. We are bound to ata186 as UA. It is compatible with this method.
Support for other UAs isn't required.
3. The calls will be terminated to Cisco GWs, while COMEDIA support
was recently added into Cisco IOS, so that theoretically the only
thing we need is to add received/rport support into proxy/registrar
and update IOS at termination points.
4. No media relay is allowed, because this will create excessive
bandwith load in a single point.
5. COMEDIA support is likely to become part of the standard, so that
our investments into development are protected.
-Maxim
On Fri, Jan 10, 2003 at 02:26:07PM +0100, Jiri Kuthan wrote:
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There is actually a plenty of options how to
traverse NATs.
Sadly, none of them works in all possible scenarios.
a) STUN -- some phones (kphone for linux, snom hardphones)
have the ability to "fool" NATs to accomplish traversal
using the STUN protocols; particularly good if you cannot
manipulate the NAT
b) geek tweaks -- you have a configurable NAT and configurable
phones (there are some of both of them). you configure static
port forwarding in the NAT and phones to advertise the
public address in contacts and elsewhere
c) ALG -- use a SIP-aware NAT such as PIX or Intertex
d) UPnP -- takes UPnP enables phones (snom is) and NATs
e) SIP/media relay -- that's a too ugly story
What to choose best depends on your network setting -- can you
tweak the NAT, can you afford replacing it with a SIP-enabled
one, are the phones you are using configurable or do they use
STUN, do you have a server on the public or private NAT side
or on each of them, etc.
I remember someone shared with us he was using ser in his
network to do the translation of SIP addresses on behalf
ot the phones. The ser script was configured to statically
rewrite private IP addresses to the public address using
replace/textops.
-Jiri
At 01:32 PM 1/10/2003, Maxim Sobolev wrote:
>Folks,
>
>I need an advise on how to better implement one feature,
which isn't
>currently present in SER. We need to allow
UAs behind NAT properly
>register with the registrar - by "properly" I mean that
host:port
portion
>of URI in Contact field should not be used,
but host:port
the request
>came from should be used instead. By
definition we know
that those UAs
>will support symmetric SIP signalling, so
that this scheme
will work just
>fine.
>
>In my opinion there are two ways to do it: either add new
rewritecontact*
>family of functions similar to rewritehost
ones. or add a
new flag for
>the save() function. This is where I need
your help -
which implementation
>looks better for you (or maybe you have even
some better
idea), since
>we are really interested in inclusion of our
changes into
the mainline to
reduce our
local hacks.
Regards,
Maxim
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Jiri Kuthan http://iptel.org/~jiri/