<div dir="ltr">Thanks Stanislav!<div><br></div><div>I understand that jitter buffer is not necessarily codec dependent and that SILK is actually the foundation of Opus, so I assume their quality is comparable.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Jan 29, 2016 at 8:26 PM, Stanislav Sinyagin <span dir="ltr"><<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">add SILK to the list (it's actually what Skype uses), and jitter buffer.<br>
<br>
but I don't have much practical experience.<br>
<br>
The quality of the sound depends also on the particular SIP client<br>
implementation.<br>
<div><div class="h5"><br>
<br>
<br>
On Fri, Jan 29, 2016 at 11:32 PM, Oleg Stolyar <<a href="mailto:olegstolyar@gmail.com">olegstolyar@gmail.com</a>> wrote:<br>
> Hi guys,<br>
><br>
> does anyone have experience using different codecs to reduce choppiness and<br>
> metallic/robotic sound when connecting to FreeSWITCH from a mobile device?<br>
><br>
> I know both OPUS and G.722 have packet loss concealment but I wonder if<br>
> someone actually has practical experience with different codecs over iffy<br>
> mobile connections.<br>
><br>
> OPUS vs. G.722 vs G.711<br>
><br>
><br>
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</blockquote></div><br></div>