On Fri, Jan 10, 2003 at 09:38:11AM -0600, Greg Fausak wrote:
Howdy,
I have a ATA186. I may have misunderstood the COMEDIA reference. Does the ATA186 poke itself through a NAT router to a SIP server???
The problem is that ATA186, while de-facto is partially COMEDIA-compatible as it always sends and receives UDP messages using the same port, i.e. 5060 for signalling and 10000 for RTP, it doesn't yet announce it to the world by inserting a blank rport parameter into the header and couldn't be configured to add direction=passive into SDP. Therefore, SIP server should be properly modified to add missed parts when it receives request from ATA.
Another problem is that those symmetric schemes don't work when both parties are behind NATs, or when the call terminates to UA which doesn't support direction=active.
-Maxim
NAT seems to be the biggest hurdle we have.
----greg
-----Original Message----- From: Maxim Sobolev [mailto:sobomax@FreeBSD.org] Sent: Friday, January 10, 2003 8:58 AM To: Greg Fausak Cc: 'Jiri Kuthan'; serusers@lists.iptel.org; kapitan@portaone.com Subject: Re: [Serusers] Rewriting URI in the Contact field
On Fri, Jan 10, 2003 at 08:27:46AM -0600, Greg Fausak wrote:
What is this device? Where can I get one? What does it cost?
Cisco ata186 is two-port analog telephone adapter, i.e. it has two standard ports for connecting ordinary phones and one 10M ethernet port. It supports SIP and H323 (G711, G723 and G729 audio codecs) and costs some US$150.
-Maxim
Thanks :-)
---greg
-----Original Message----- From: serusers-admin@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@lists.iptel.org; kapitan@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:
- Things should work without modifying or reconfiguring existing
user's infrastructure (NATs) and should be compatible with all widely-used NATs.
- We are bound to ata186 as UA. It is compatible with
this method.
Support for other UAs isn't required.
- 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.
- No media relay is allowed, because this will create excessive
bandwith load in a single point.
- 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:
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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Jiri Kuthan http://iptel.org/~jiri/
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