[COLUG] Asterisk

[Duane]

Rob Funk wrote:
> Duane wrote:
>> Aaron Howard wrote:
>>> Does anybody on the list have actual production implementation
>>> experience w/ Asterisk?
>> Don't do it, it's not worth it! :)
>>
>> If you have to use Asterisk, use SER/OpenSER in front of Asterisk.
> 
> Why?  What's wrong with Asterisk, and why is OpenSER better?
> 
> I intend to set up VoIP between home and work soon, and I was planning on 
> using Asterisk, partly because of the IAX protocol's NAT-traversing 
> advantage over SIP.

There isn't really an advantage if setup correctly, you just need to
specify outbound proxy or use STUN, I have used outbound proxies to go
through 2 NAT layers without any problems what so ever.

As for what's wrong with Asterisk, it just doesn't scale, it has a
poorly implemented SIP stack, and any commercial provider that runs it
as the primary VoIP platform suffers as a result, and usually needs a
LOT LOT more hardware then companies using other platforms.

SER/OpenSER and friends unfortunately are a bit black magicish when it
comes to configuring them, but once you have it setup, combined with
RTPProxy it can actively detect if you are behind nat and deal with it
accordingly, so part of the "NAT-traversing advantage" IAX has over SIP
is purely due to poorly implemented SIP code in Asterisk.

Simply put, if you want to scale and you want to have SIP "just work"
then it can be done even with Asterisk.

Callweaver is a fork of Asterisk and while they seem to have made
numerous improvements in Asterisk code, unfortunately they didn't get
round to integrating with sofia which is a replacement SIP stack, and
now because of feature freeze going into 1.0 release they most likely
won't for a while.

Asterisk coders tend to have a habit of closing bugs without caring if
problems have been fixed and it's just horrid code in places, take the
way Asterisk gets an 8k timing source, they flood the PCI bus with
interrupt requests *shudder*

Callweaver guys are mostly former Asterisk guys disgruntled by digium's
business practices getting in the way of better code, oh and if you need
fax/t.38 etc, Callweaver comes with it by default.

Also in this day and age of fibre trunks having packets copied to every
LEA out there something like advantageous encryption is a must,
unfortunately neither Asterisk nor Callweaver seem to do this yet, with
sofia code Callweaver might, SER/OpenSER on the other hand have had this
coded for quite some time now, you need to run SIP over TCP for the
channeling/data commands to be able to easily exchange keys, Asterisk
has never had TCP SIP support.

I could go into how bad the enum lookup stuff in Asterisk is, and how
the freepbx guys got sick of complaints and wrote their own lookup
function in PHP instead, or how Asterisk has had a long list of buffer
overflow bugs, or how every Asterisk release usually has a number of
critical security flaws, but I think I should stop now :)

-- 

Best regards,
 Duane

http://www.freeauth.org - Enterprise Two Factor Authentication
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"In the long run the pessimist may be proved right,
    but the optimist has a better time on the trip."