<br><br><div class="gmail_quote">On Tue, Nov 18, 2008 at 11:00 PM, mark morreny <span dir="ltr"><<a href="mailto:markmorreny@gmail.com">markmorreny@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>Hi,</div>
<div> </div>
<div>I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. </div>
<div> </div>
<div>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.</div>
<div> </div>
<div>Does anyone know why? Does it have anything to do with what codec to use?</div>
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<div>Thanks, </div>
<div>Mark</div>
<br></blockquote></div>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. <br>
<br>Thanks & Regards,<br>Godson Gera<br><a href="http://godson.in">http://godson.in</a><br>