<div dir="ltr"><div>probably the best you can do is limit the number of transcodings and/or resamplings</div><div><br></div><div><div>skype uses silk</div><div><span class="" style="white-space:pre">        </span>freeswitch core uses L16</div><div><span class="" style="white-space:pre">                </span>sip session uses <your choice></div><div><span class="" style="white-space:pre">                        </span>asterisk core uses ?</div><div><span class="" style="white-space:pre">                                </span>audio is presented to the dongle as ?</div><div><span class="" style="white-space:pre">                                        </span>cellular network uses gsm</div><div><br></div><div>if ? = L16 then make the sip session use L16</div><div>use the loopback interface so that MTU is not an issue</div></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi <span dir="ltr"><<a href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>for this setup<br></div>skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine <br><br>suggest best codecs to use with freeswitch and asterisk for good voice quality ??<br></div>
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