[Asterisk-Users] SER & Asterisk combination to get around NAT

[Stuart Hirst]

Mark,

Thanks for your response.

The typical deployment is a single server in the customer location
directly on the end of an ADSL link with two Ethernet interfaces, 1 to
the ADSL modem and the other to the LAN. The LAN side is fine and is as
normal but many customers have remote users or remote small offices that
may have more than one SIP device behind NAT.

What I am trying to establish is how successful SER is at allowing
multiple remote SIP devices behind a remote NAT router to interact with
Asterisk and what issues need to be taken into account such as MWI and
or codec's.

I have been using Asterisk for quite some time but have not played with
SER yet and so does anyone have some sample SER configs to work in this
type of deployment.

Stuart


-----Original Message-----
From: Mark John Buenconsejo [mailto:mjwork at gmail.com] 
Sent: 18 November 2005 06:32
To: stuart.hirst at holdentel.com
Subject: Re: [Asterisk-Users] SER & Asterisk combination to get around
NAT
Importance: High


Hello Stuart, we have, and I would be happy to help you setup both
Asterisk and SER on a consultancy basis.

You can find more information about me here:
http://mark.teamcebu.com

Basically, it requires SER to forward the SIP messages to Asterisk, and
that SER be configured as one of the SIP channels on Asterisk. What
happens is:

from the LAN Phone, it connects to SER 
and then SER forwards it to Asterisk 
Asterisk will connect to the actual destination 
As soon as Asterisk is able to connect to the destination, it then
replies to the phone that the call is connected 
At this point, the actual call connections are made (asterisk-to-phone
and asterisk-to-destination)

and then Asterisk bridges the asterisk-to-destination and
asterisk-to-phone connections 
The bridged call mechanism on Asterisk works around the NAT limitations 
In this setup, it will appear that the Phone is connecting to Asterisk
(LAN side), and that the destination is talking to Asterisk (Live side),
and Asterisk passes the RTP packets back-and-forth.

There are a few considerations though, such as codec supports. As much
as possible use the same codec for each leg of the call, otherwise the
call quality deteriorates during transcoding.

By the way, we're using this with up to 12 simultaneous calls in our
setup (a small call center), using either iLBC and G.729 codec.

Anyway, let me know if you need further help. :) Or if you have some
more specific questions.

Thanks!

Mark

Stuart Hirst wrote: 
Has anyone successfully used SER and Asterisk together on the same
server to get around NAT traversal issues.

I have looked at many of the NAT traversal topics which either involve
commercial products and significant costs or solutions such as STUN or
proprietary systems such as xten.

  



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