[asterisk-users] sound quality between two back-to-back asterisk

[Godson Gera]

On Tue, Nov 18, 2008 at 11:00 PM, mark morreny <markmorreny at gmail.com>wrote:

> Hi,
>
> I have two asterisks that are connected to each other via a back-to-back E1
> link using a pair of sangoma cards.
>
> With the following scenario:  SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk
> <-> SIP-PHONE, the sound quality degrades significantly.   I can't
> understand why as the amound of packet lost should be very minimum.
>
> Does anyone know why?  Does it have anything to do with what codec to use?
>
> Thanks,
> Mark
>
> what is the sip phone that you are using ? is it a IP phone instrument or a
softphone ? Try running with ulaw or alaw (g711) . Cause we found that
certain softphones with gsm or other codecs like speex can produce really
bad audio.

Thanks & Regards,
Godson Gera
http://godson.in
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