[Asterisk-biz] www.google.com/talk/

[Rehan Ahmed AllahWala - Super Technologies I]

Dear Ariel and others,

Currently i think we should all think of how we can use it, or start a new open source 
softphone project that is as small as this software, and as good as it is.

It has no echo what so ever like in skype.

Xten has done a great job on letting service providers use sip, but it does not have 
the echo cancellation like in skype.

So who will be the next mark spencer for the sip softphone ?

Rehan




> That is actually the price, maybe it's not that much for Google. Or
> they know something we don't.
> 
> Ariel Mónaco - Systems Engineer
> Flylabs.com - Communications and Information Technology
> 
> Potosi 4456 12th floor "D"
> Ciudad Autónoma de Buenos Aires (1199)
> Buenos Aires - Argentina
> +54-11-4983-8142
> arielmonaco at flylabs.com
> ----- Original Message ----- 
> From: "Jonathan k. Creasy" <jonathan at bluegrass.net>
> To: "Commercial and Business-Oriented Asterisk Discussion" 
> <asterisk-biz at lists.digium.com>
> Sent: Thursday, August 25, 2005 3:00 PM
> Subject: RE: [Asterisk-biz] www.google.com/talk/
> 
> 
> Another thought btw, who says they are paying those prices? Couldn't
> they just negotiate a deal say "$1M for all our users" and be done
> w/it?
> 
> -Jonathan
> 
> -----Original Message-----
> From: asterisk-biz-bounces at lists.digium.com
> [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Eric
> Wieling aka ManxPower Sent: Thursday, August 25, 2005 1:06 PM To:
> Commercial and Business-Oriented Asterisk Discussion Subject: Re:
> [Asterisk-biz] www.google.com/talk/
> 
> Some?
> 
> Here is the licensing priceing info for G723.1 direct from the patent
> holder's web site: http://www.dspg.com/technology/LicensePricing.html
> 
> 
> 
> Jonathan k. Creasy wrote:
> > They probably don't have to make money on it at this point....they
> > can afford to lose some to get the momentum they need to make money
> > on it later... -Jonathan
> >
> > -----Original Message-----
> > From: asterisk-biz-bounces at lists.digium.com
> > [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Eric
> Wieling
> > aka ManxPower
> > Sent: Thursday, August 25, 2005 1:00 PM
> > To: Commercial and Business-Oriented Asterisk Discussion
> > Subject: Re: [Asterisk-biz] www.google.com/talk/
> >
> > Steve Kennedy wrote:
> >
> >
> >>Which voice codecs do you support?
> >>
> >>Today, Google Talk supports the following standard voice codecs:
> >>PCMA, PCMU, G.723, iLBC. We are also evaluating the Speex codec. We
> >>also support codecs from Global IP Sound: ISAC, IPCMWB, EG711U,
> >>EG711A
> >
> >
> > I'd love to know how they support G723 in a free application.
> > _______________________________________________
> > Asterisk-Biz mailing list
> > Asterisk-Biz at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-biz
> > _______________________________________________
> > Asterisk-Biz mailing list
> > Asterisk-Biz at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-biz
> >
> 
> 
> -- 
> Eric Wieling * BTEL Consulting * 504-210-3699 x2120
> 
> r: Generate a ringing tone for the calling party, passing no audio
> from the called channel(s) until one answers. Use with care and don't
> insert this by default into all your dial statements as you are
> killing call progress information for the user. Really, you almost
> certainly do not want to use this. Asterisk will generate ring tones
> automatically where it is appropriate to do so. "r" makes it go the
> next step and additionally generate ring tones where it is probably
> not appropriate to do so.
> 
> _______________________________________________
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> Asterisk-Biz at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-biz
> _______________________________________________
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> Asterisk-Biz at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-biz
> 
> _______________________________________________
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> Asterisk-Biz at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-biz

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