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

[John Todd]

>-----Original Message-----
>From: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of John Todd
>Sent: Monday, October 13, 2003 8:11 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
>
>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

At 11:26 PM -0400 10/13/03, Uriel Carrasquilla wrote:
>From: "Uriel Carrasquilla" <uriel at adelphia.net>
>To: <asterisk-users at lists.digium.com>
>Subject: RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP 
>Phones on the Internet)
>Reply-To: asterisk-users at lists.digium.com
>Date: Mon, 13 Oct 2003 23:26:59 -0400
>
>John:
>are you aware of any documentation on how to configre SER to be a front-end
>to Asterisk?
>I suspect it is very inexpensive to put a SER server in a hosting facility
>to forward traffic to multiple Asterisks based on Least Cost Routing.
>My problem is that my experience is with Asterisk and not with SER.
>  Uriel

Uriel -
   1) Please stop top-posting.

   2) I'm afraid I don't have any data on specifics of creating a 
front-end.  I know how to do it, but my time these days is spent 
writing lots of other projects that I have been doing.  :-)  I would 
suggest you get SER and set it up - it's quite easy, and the 
documentation on SER itself is very well written, and if you have a 
good idea of how SIP works you should be able to patch together an 
appropriate system.  However, if you aren't 100% familiar with how 
SIP works, I would stick to just an Asterisk system; SER doesn't 
allow for any of the "shortcuts" that Asterisk has.

   3) Use Google and do some searching.  I found some quick links with 
a few of the keywords that would seem obvious, but I don't have 
enough time to review them...

JT