Im trying to write some tutorial for these ever recurring SIP + NAT
questions.
Its far from ready, and its without layout, but the draft can be found
at: http://www.asteriskguru.com/natut.php it has all most of the
situations explained, and explains all the options you need to look at
in the asterisk config files.
/Z
Olle E. Johansson wrote:
Irakli Natsvlishvili wrote:
100k question - does asterisk correctly handle following situations:
There are plenty of good documents on Asterisk, SIP and NAT on the
voip-info.org wiki. Please look them up. There are also information
within the configs/sip.conf.sample file within Asterisk.
1. Asterisk is on a public IP
Two SIP clients on separate networks, each of them are behind
dynamic NAT
gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go
thought
asterisk.
If the media stream SHOULD NOT go through Asterisk, then it's up to
the phones
to support NAT traversal properly and handle this, it's not an
Asterisk problem. From Asterisk's point of view, we should not see
that they are in fact behind NAT. Modern phones in combination with
STUN and a decent NAT device supports this.
2. Even worst case - three clients, two of them on one site, second
is on
another site. For example extensions 500 and 600 are on the same site
and in
the same subnet and extension 1000 is on another site/network. There
are PAT
FW/gateways with dynamic public IP in front of clients and those are
symmetric NAT/FW.
The task - clients registering on Asterisk server, calling each other
and
RTP should not go via asterisk. So, media stream should go directly
from one
client to another.
If Asterisk is on a public IP, again: it's up to the phones. It's
still not
an Asterisk problem.
I want to know:
1. Is it possible? - yes/no. Implementation should involve asterisk
and SIP
clients and not involving third party hardware products - ALG, session
border controllers or so on.
Yes, but you need to pick the right phone, the right NAT/FW and have a
lot of patience :-)
2. If it is possible, what are requirements for SIP clients.
Good NAT traversal support.
3. What configuration changes should be done on Asterisk server and
on a sip
clients.
From Asterisk's point of view, all of these phones are on a public IP
and we
do not give them any NAT traversal support. If you want detailed
configurations, there are several consultants available that can help
you with that (including my company).
And final question - if it is NOT possible with Asterisk, do you know an
open source product which works in above stated scenarios and you've
actually tested it.
It is possible with Asterisk and every other SIP server. With your
requirements, it's completely a client-side problem.
Best regards,
/Olle
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