[Asterisk-Users] iconnect quality?

Krzysztof Bujak kbujak at itform.pl
Mon Mar 10 17:52:18 MST 2003


Dan, welcome to the club.
Country like Poland which is supposed to get into EU is also 3rd world in
this matter.
(However things are changing a lot lately)

Regards,
Krzysztof Bujak

----- Original Message -----
From: "Dan Fernandez" <danfernandez00 at hotmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, March 10, 2003 11:18 PM
Subject: Re: [Asterisk-Users] iconnect quality?


> Yes, 64K is not much of a broadband and believe it or not, I am paying
US$60
> for it (I believe this is the case in many 3rd world countries).
>
> Is there a codec translator between GSM and g723?
>
> How come I can use FWD just fine with g711?
>
> ----- Original Message -----
> From: "Steven Critchfield" <critch at basesys.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Monday, March 10, 2003 5:33 PM
> Subject: Re: [Asterisk-Users] iconnect quality?
>
>
> > On Mon, 2003-03-10 at 13:47, Dan Fernandez wrote:
> > > Iconnect uses codecs g723 and g711 that can be configured for each
> account
> > > (you can change them by the 7777 prefix)
> > >
> > > With their dialer and g723 I can here just fine (I  have a 64k
broadband
> > > connection). With their dialer and g711 the quality suffers greatly.
> > >
> > > With * and GSM I cannot here anything (and don´t know if they can here
> me).
> > > The call gets logged on  iconnect´s CDR. Upon looking at the SIP debug
> > > everything appears just fine, but again, I cannot hear anything.
> >
> > g711 is 8 bit 8khz, or 64Kbit of audio data alone without the overhead
> > of TCP/IP nor SIP. 64Kbit is not broadband, it is a DS0. It may be a tad
> > over dial up, but please don't consider it broadband.
> >
> > This would explain your quality problem, you fill the link in the first
> > set of samples, and the rest is queued up and therefore does not arrive
> > like a stream should. Each fram probably adds a millisecond or more to
> > the stream and soon enough you are so starved for audio data that it
> > should give up.
> > --
> > Steven Critchfield  <critch at basesys.com>
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > 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
>
>
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