Hello,<div><br></div><div>I am currently using the following setup:</div><div>Snom 300 - Asterisk 1.8.13.0 running on Raspberry Pi - Sipgate SIP Provider</div><div>When I am using this setup, the call quality isn't as good as when using a direct connection like</div>
<div>Snom 300 - Sipgate SIP Provider
</div><div>to my SIP Provider (Sipgate).</div><div><br></div><div>Sipgate supports</div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<ul style="font-family:arial,tahoma,verdana,sans-serif;font-size:13px;line-height:20px"><li>G.729 - ~ 12 kbit/s</li><li>G.711 - ~ 100 kbit/s</li><li>iLBC - ~ 15 kbit/s</li><li>GSM - 13 - 20 kbit/s<br></li><li>G.726 - 16 - 40 kbit/s</li>
</ul></blockquote><div>According to my Asterisk server and sip show channels the used codec is 0x08 (<span style="font-family:arial,tahoma,verdana,sans-serif;font-size:13px;line-height:20px">G.711), so the best one possible. Still the quality isn't as good as on a direct connection between my Snom phone and Asterisk. Using Asterisk it feels a little bit like there is some more background noise or so. Any hints why the quality using Asterisk is less good than on a direct connection? Does Asterisk by default change anything on the signal or is it just passed through? Thanks :-)</span></div>
<div><span style="font-family:arial,tahoma,verdana,sans-serif;font-size:13px;line-height:20px"><br></span></div><div><span style="font-family:arial,tahoma,verdana,sans-serif;font-size:13px;line-height:20px">Best regards</span></div>
<div><font face="arial, tahoma, verdana, sans-serif"><span style="line-height:20px">Stefan</span></font></div>