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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>If you are using mobile data to set up your own SIP calls, then you are now struggling through the same battles that the cellular carriers conquered for the past 2 decades. </span><span style='font-size:11.0pt;font-family:Wingdings;color:#1F497D'>L</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> LTE should help greatly with this, as the carriers in the U.S. are moving to send everything over LTE.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>If it is a mobile voice call, then you are at the mercy of the codec audio that the cellular carrier chose.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Bote<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt'><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Volodymyr Fedorov<br><b>Sent:</b> Saturday, 30 January, 2016 12:27<br><b>Subject:</b> Re: [Freeswitch-users] Audio Codecs over Mobile Connection<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Question is what kind of mobile connection do you use? If it is UMTS/CDMA, so even not compressed codecs like g711u/a or g722 are working fine. Regarding bandwidth usage you have to choose between SILK,Opus or even old one iLBC or GSM.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div></div></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Sat, Jan 30, 2016 at 9:40 AM, Stanislav Sinyagin <<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>> wrote:<o:p></o:p></p><p>Of course jitter buffer is independent from codec, but its functionality affects the quality.<o:p></o:p></p><p>Yes SILK is inherited by Opus, but in general SILK is much cheaper in terms of CPU cycles for encoding and decoding.<o:p></o:p></p><p>You can see the difference here: <a href="https://txlab.wordpress.com/2015/04/10/testing-freeswitch-performance-on-scaleway-c1/" target="_blank">https://txlab.wordpress.com/2015/04/10/testing-freeswitch-performance-on-scaleway-c1/</a><o:p></o:p></p><div><div><div><p class=MsoNormal>On 30 Jan 2016 06:41, "Oleg Stolyar" <<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>> wrote:<o:p></o:p></p><div><p class=MsoNormal>Thanks Stanislav!<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>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.<o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Fri, Jan 29, 2016 at 8:26 PM, Stanislav Sinyagin <<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>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.<o:p></o:p></p><div><div><p class=MsoNormal><br><br><br>On Fri, Jan 29, 2016 at 11:32 PM, Oleg Stolyar <<a href="mailto:olegstolyar@gmail.com" target="_blank">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>><o:p></o:p></p></div></div></div></div></div></div></div></div></div></div></div></body></html>