<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:16px"><div id="yui_3_16_0_1_1418724125078_4987" dir="ltr"><span>Hi</span></div><div id="yui_3_16_0_1_1418724125078_4987" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1418724125078_4987" dir="ltr"><span id="yui_3_16_0_1_1418724125078_7682">You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if there is packet loss. You know if there's packet loss if you receive RTCP . </span></div><div id="yui_3_16_0_1_1418724125078_4987" dir="ltr"><span>Linphone does this .</span></div><div id="yui_3_16_0_1_1418724125078_4987" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1418724125078_4987" dir="ltr"><span>Regards,</span></div><div id="yui_3_16_0_1_1418724125078_4987" dir="ltr"><span>Dragos Oancea</span></div><br> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;" id="yui_3_16_0_1_1418724125078_4991"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;" id="yui_3_16_0_1_1418724125078_4990"> <div dir="ltr" id="yui_3_16_0_1_1418724125078_4989"> <hr size="1" id="yui_3_16_0_1_1418724125078_7681"> <font size="2" face="Arial" id="yui_3_16_0_1_1418724125078_4988"> <b><span style="font-weight:bold;">From:</span></b> Manpreet Singh <manpreets7@gmail.com><br> <b><span style="font-weight: bold;">To:</span></b> opus@xiph.org <br> <b><span style="font-weight: bold;">Sent:</span></b> Tuesday, December 16, 2014 7:54 AM<br> <b><span style="font-weight: bold;">Subject:</span></b> [opus] Estimating bitrate during a real-time voip call<br> </font> </div> <div class="y_msg_container" id="yui_3_16_0_1_1418724125078_4993"><br><div id="yiv6635412837"><div dir="ltr" id="yui_3_16_0_1_1418724125078_4994">Hi,<div id="yui_3_16_0_1_1418724125078_7680"><br></div><div id="yui_3_16_0_1_1418724125078_4995">Although this maybe considered out of scope here, but I'll ask anyway.</div><div id="yui_3_16_0_1_1418724125078_7679"><br></div><div id="yui_3_16_0_1_1418724125078_4996">Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly?</div><div id="yui_3_16_0_1_1418724125078_7678"><br></div><div id="yui_3_16_0_1_1418724125078_4997">Thanks,</div><div id="yui_3_16_0_1_1418724125078_7677">Manpreet.</div><div id="yui_3_16_0_1_1418724125078_4998"><br></div></div></div><br>_______________________________________________<br>opus mailing list<br><a ymailto="mailto:opus@xiph.org" href="mailto:opus@xiph.org">opus@xiph.org</a><br><a href="http://lists.xiph.org/mailman/listinfo/opus" target="_blank" id="yui_3_16_0_1_1418724125078_7683">http://lists.xiph.org/mailman/listinfo/opus</a><br><br><br></div> </div> </div> </div></body></html>