[asterisk-users] Platform decision ...

[Steve Totaro]

On Tue, Aug 18, 2009 at 1:52 PM, Mauro Sergio Ferreira Brasil <
mauro.brasil at tqi.com.br> wrote:

> Hello there!
>
> During some research on Internet I found the following comparison on
> site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ
> "):
>
> The main points listed on Asterisk's "CONS" that concerned me were:
>
>   * Conferencing on Asterisk depends on Zaptel hardware and/or kernel
> modules for timing;
>   * Lack of built-in STUN support for SIP NAT traversal;
>   * Asterisk doesn't use SpanDSP;
>   * Use of no longer maintained Berkeley DB1 engine as its internal
> database;
>   * Asterisk doesn't allow CSRC entries in RTP;
>   * Asterisk doesn't have an universal jitterbuffer for use with any
> channel type;
>   * Asterisk doesn't use POSIX realtime extensions (having dependency
> with Zaptel timing);
>
> We were considering Asterisk as the chosen platform, but after reading
> this I got a little worried.
> The comparison considers 1.4 old version of Asterisk.
>
> So, can someone give me an update on what have changed for this items
> considering new 1.6 version ?
> Maybe someone can point me a site with an updated comparison.
>
> As long as I could see by now SpanDSP is present on new version of
> Asterisk, so this item isn't a difference any more. Right ?
>
> Thanks and best regards,
>
> --
> __At.,
>   _
>
> *Technology and Quality on Information*
> Mauro Sérgio Ferreira Brasil
> Coordenador de Projetos e Analista de Sistemas
> + mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
> : www.tqi.com.br <http://www.tqi.com.br>
> ( + 55 (34)3291-1700
> ( + 55 (34)9971-2572
>
>
Don't forget to add FreeSwitch to your comparison chart too.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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