[Asterisk-Users] Asterisk VS. Skype
Mathieu Nantel
mnantel at teclinux.com
Thu Apr 29 17:32:36 MST 2004
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
> > _______________________________________________
> > 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
> >
>
> _______________________________________________
> 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
>
> _______________________________________________
> 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
>
>
--
Mathieu Nantel, RHCE
TecLinux.com
mnantel at teclinux.com
More information about the asterisk-users
mailing list