[Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

[Uriel Carrasquilla]

Chris:
I am glad to see someone else asking the same question I have been asking
myself.
As soon as I get my public IP address, I will install SER on the public side
and Asterisk behind a NAT (with dynamic IP) to see if I can get around
problems I have when my SIP (UA) behind their own NAT on the other side of
my Internet connection.
If you make any progress, please share.  I will do the same.
Uriel

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Chris
Albertson
Sent: Monday, October 13, 2003 7:49 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)



I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.

SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route based on what it finds or even
re-write packets based on user specified logic.

SER is GPL'd and has very good user documentation.  Don't know how well
the above will work.  The claim by the authors or SER that it can
handle thousands of calls per second is quite impressive

One other nice feature is that SER users can set up their own SIP
accounts using a web interface and not needing  to edit *.conf files.

See here for details http://www.iptel.org/ser/


=====
Chris Albertson
  Home:   310-376-1029  chrisalbertson90278 at yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson at aero.org
  KG6OMK

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