[Asterisk-Users] one way choppy sound problem !

TeleSIP ricvil at telesip.net
Mon Jan 5 11:21:33 MST 2004


Wipeout,  If you want you can send me an Ethereal trace of the RTP stream
and I can do an analysis of it to determine if there is anything obvious
there. (please use G.711, and try something like counting from 1 to 20).

Regards,
Andres

----- Original Message ----- 
From: "WipeOut" <wipe_out at users.sourceforge.net>
To: <asterisk-users at lists.digium.com>
Sent: Monday, January 05, 2004 11:22 AM
Subject: Re: [Asterisk-Users] one way choppy sound problem !


> Michael Van Donselaar wrote:
>
> >On Mon, 5 Jan 2004 13:29:06 +0100, "Dawid Mielnik"
<D.Mielnik at elka.pw.edu.pl>
> >wrote:
> >
> >
> >
> >>Hi Again,
> >>
> >>Apart from X-lite client I have also tried eStara, diax phone, iaxcomm
and
> >>some others. I have tried different codecs - GSM, aLAW uLAW. They all
give
> >>the same result. In the direction PSTN user ---> Softphone user the
sound is
> >>crystal clear (also tried on a dial-up connection), in the other
direction
> >>however the sound is a bit choppy. The chops occur at regular intervals
of
> >>time, at about 1-2 seconds !?
> >>
> >>
> >
> >Are the PSTN interface and a network card sharing an interrupt?  I had
similar
> >problems with my X100P and a thunderlan dual ethernet card shring IRQs
(also
> >would make one of the ethernet ports fails until reboot)
> >
> >Are you still using the P133?  I tried using a P120, but it wouldn't do
the
> >trick with GSM conversion.  DIAX and iaxComm, since they use the
iaxclient
> >library, need to use GSM.
> >
> >
>  I have the same choppy sound problem on my server, my card is not
> sharing an interrupt and I am using G711 which is not hittng the P2 400
> at all.. It seems there is a gremlin.. :)
>
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> Asterisk-Users at lists.digium.com
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>





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