[Asterisk-Users] Asterisk VS. Skype

[brian]

The firefly ulaw codec isn't working right with the latest CVS code... I
ment to bring that up to adam.

bkw

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Mathieu Nantel
> Sent: Thursday, April 29, 2004 7:33 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Asterisk VS. Skype
>
> You know what: I just figured out my problem. I was using Firefly on one
> end and IAX Phone at the other end. And guess what, IAX Phone doesn't
> support G711, yielding to mediocre quality and latency for what I
> suppose was translating G711 <-> GSM
>
> > >>B) Asterisk won't be adding latency on LAN->LAN, it's your end clients
> -
> > try Firefly - www.virbiage.com (insert bias comment here) and see if
> > >>that's addresses your issues.
> > In my (limited) experience I've found softclients always produced poor
> > results (choppy lagged voice); CISCO-ATA and CISCO7960 however performed
> > perfectly and usually better than PSTN call quality.  I spent two hours
> > chatting to my friend in the US (I live in the UK) and he was stunned
> at the
> > end of the call when I told him it was a voip->pstn call via voicepulse.
> >
> >
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Adam Hart
> > Sent: 30 April 2004 00:23
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] Asterisk VS. Skype
> >
> > A) Skype uses iLBC as a codec which Asterisk already has
> > B) Asterisk won't be adding latency on LAN->LAN, it's your end clients
> - try
> > Firefly - www.virbiage.com (insert bias comment here) and see if that's
> > addresses your issues.
> >
> >
> > Mathieu Nantel wrote:
> > > This might have been talked about before, but I'm posting anyhow.
> > >
> > > I've got down to testing Asterisk yesterday, and I couldn't help but
> > > compare it with Skype (a Windoze only product, yet, but extremely
> > > efficient for some reason).
> > >
> > > Skype has almost unperceptible delay (LAN), while there is almost half
> > > a second of delay regardless of the codec on Asterisk.
> > >
> > > An even if we were to eliminate the latency on asterisk, Skype has at
> > > least 2x the quality of PSTN, let alone any codec on asterisk.
> > >
> > > So I ask: are there no royalty-free codecs that come close to those
> > > from GIPS? (I understand these people are supplying the codecs for
> Skype).
> > >
> > > Or is there something I'm doing wrong? What configuration yields the
> > > highest quality and the lowest bitrate? I'm talking LAN communication
> > > here: plenty of bandwidth, no latency.
> > >
> > > (I'm aware that this could be a flamebait. Bring em on.)
> > >
> > > --
> > > Mathieu Nantel
> > > TecLinux.com
> > > mnantel at teclinux.com
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>
> --
> Mathieu Nantel, RHCE
> TecLinux.com
> mnantel at teclinux.com
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