[Asterisk-Users] Re: g729 to asterisk to g729 voip provider

[Christian B]

On Fri, 16 Sep 2005 16:09:37 -0500
Erick Perez <eaperezh at gmail.com> wrote:

> Hi, your project is indeed interesting, however for learning purposes
> i do need to know the answer of at least:

it is not my project.

> 
> 1- Using sipura sip/g729 to connect to an asterisk server that will
> server as a gateway to a VOIP provider(g729), all in g729 will require to
> purchase codecs from Digium?

read the page, it provides you with a free version of the g729

> 
> 2- also, in this scenario the transcoding is almost non-existent right?

since all codecs are the same, nothing has to be transcoded of course.

> 
> 3- I have read many documents about the type of codecs, and g729 seems to
> be a good trade between almost-toll quality and low bandwith usage
> right?

yes, it offers a good trade between both. however, voice quality is not superb but satisfying.

regards
christian


> On 9/16/05, ChB <a0305292 at unet.univie.ac.at> wrote:
> > Hello Erik!
> > 
> > check out this website: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
> > i have both codecs running on gentoo without problems(working with cisco 7960 and snom 190). regarding your hardware question i don't have enough experience yet, sorry.
> > 
> > regards
> > christian
> > 
> > 
> > On Fri, 16 Sep 2005 15:01:32 -0500
> > Erick Perez <eaperezh at gmail.com> wrote:
> > 
> > > anyone with some info on this?
> > >
> > > thanks again.
> > >
> > > On 9/14/05, Erick Perez <eaperezh at gmail.com> wrote:
> > > > Using sipura sip/g729 to connect to an asterisk server that will
> > > > server as a gateway to a VOIP provider, all in g729 will require to
> > > > purchase codecs from Digium?
> > > >
> > > > also, in this scenario the transcoding is almost non-existent right?
> > > > I have read many documents about the type of codecs, and g729 seems to
> > > > be a good trade between almost-toll quality and low bandwith usage
> > > > right?
> > > >
> > > >
> > > > A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can
> > > > sustain more than 100 calls or up to a 100?
> > > > I just looking at hardware capacity, since the machine will be located
> > > > at an ISP with more than needed bandwith.
> > > >
> > > > There is no need for voicemail, web interfaces or anything else, since
> > > > the * box will only function as a gateway to a US-based VOIP provider.
> > > >
> > > > The machine in question runs Centos4 Linux (Redhat enterprise 4) and
> > > > CDR logging only.
> > > >
> > > > Thanks,
> > > >
> > >
> > >
> > > --
> > >
> > > -------------------------------------------
> > > Erick Perez
> > > Linux User 376588
> > > http://counter.li.org/  (Get counted!!!)
> > > Panama, Republic of Panama
> > > _______________________________________________
> > > --Bandwidth and Colocation sponsored by Easynews.com --
> > >
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > 
> 
> 
> -- 
> 
> -------------------------------------------
> Erick Perez
> Linux User 376588
> http://counter.li.org/  (Get counted!!!)
> Panama, Republic of Panama
>