On Tue, Nov 18, 2008 at 11:00 PM, mark morreny <[EMAIL PROTECTED]>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
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
