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

[John Todd]

>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

SER is an excellent option as a front end to Asterisk.  It is a 
"true" SIP proxy, whereas Asterisk is a hybrid, and SIP has not been 
the primary focus of Asterisk development.  In fact, Asterisk's SIP 
implementation is very limited (though it is extremely pragmatic.)

However, moving to SER does not solve any of the issues about the 
proxy being behind a NAT, and I believe that SER will have the same 
problems (though I could be wrong on this; I haven't experimented 
with SER's ability to work from behind a NAT.)   SIP clients work 
well enough behind NAT (most of them, anyway) but the servers are a 
different story.

I really like SER's third-party addons for account administration; 
Asterisk is significantly more complex, and probably would not be as 
easily converted to such a front end.  In fact, SER has a very 
complex routing/scripting language that is not easily administered 
with a web front end, so I think that SER and Asterisk suffer from 
the same problems.  If someone were to come up with a simple way to 
administer voicemail.conf and sip.conf from a web tool, that would go 
far to making Asterisk a bit more user-accessible...

JT