<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px"><div id="yui_3_16_0_1_1416469943651_2403">Hi,</div><div id="yui_3_16_0_1_1416469943651_2404" dir="ltr">I could to solve this issue. someone that had this problem said that jssip should run on Apache, so I did my test with sipml5 and now call will be established but there is no voice (RTP) and after 30 sec call terminated that I think is because for RTP timeout.</div><div id="yui_3_16_0_1_1416469943651_2461" dir="ltr">So are there any configs that I should do ?</div><div id="yui_3_16_0_1_1416469943651_2462" dir="ltr">Thanks.</div><div id="yui_3_16_0_1_1416469943651_2463" dir="ltr"><br></div><div id="yui_3_16_0_1_1416469943651_2464" dir="ltr"><br></div><div id="yui_3_16_0_1_1416469943651_2465" dir="ltr"><br></div><div id="yui_3_16_0_1_1416469943651_2466" dir="ltr">Regards,</div><div id="yui_3_16_0_1_1416469943651_2467" dir="ltr">H.Yavari<br></div><div id="yui_3_16_0_1_1416469943651_2400"><span></span></div><br> <div id="yui_3_16_0_1_1416469943651_2457" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;"> <div id="yui_3_16_0_1_1416469943651_2456" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1416469943651_2458" dir="ltr"> <hr size="1"> <font id="yui_3_16_0_1_1416469943651_2459" face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> H Yavari <hyavari@rocketmail.com><br> <b id="yui_3_16_0_1_1416469943651_2634"><span id="yui_3_16_0_1_1416469943651_2633" style="font-weight: bold;">To:</span></b> "spce-user@lists.sipwise.com" <spce-user@lists.sipwise.com> <br> <b id="yui_3_16_0_1_1416469943651_2636"><span id="yui_3_16_0_1_1416469943651_2635" style="font-weight: bold;">Sent:</span></b> Thursday, 20 November 2014, 10:12:08<br> <b id="yui_3_16_0_1_1416469943651_2632"><span id="yui_3_16_0_1_1416469943651_2631" style="font-weight: bold;">Subject:</span></b> Re: [Spce-user] webRTC in production<br> </font> </div> <div id="yui_3_16_0_1_1416469943651_2455" class="y_msg_container"><br><div id="yiv8708658324"><div id="yui_3_16_0_1_1416469943651_2454"><div id="yui_3_16_0_1_1416469943651_2453" style="color:#000;background-color:#fff;font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18533"><span>Hi,</span></div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18582"><span id="yiv8708658324yui_3_16_0_1_1416396176084_18581">I installed the m3.6.1 and now I can registered my sip user from browser. But now when I create a call, I receive "User Denied Media Access"</span></div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18590">and call not established.</div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18593"><br clear="none"></div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18606">I changed the "srtp_transcoding" to Force RTP but error not changed. <br clear="none"></div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18622">How can I solve this issue?</div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18612">Thanks.</div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18613"><br clear="none"></div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18615"><br clear="none"></div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18617">Regards,</div><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18619">H.Yavari<div class="qtdSeparateBR"><br><br></div><div class="yiv8708658324yqt1458283842" id="yiv8708658324yqtfd94043"><br clear="none"></div></div><div class="yiv8708658324yqt1458283842" id="yiv8708658324yqtfd79698"><div id="yiv8708658324yui_3_16_0_1_1416396176084_18378" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div id="yiv8708658324yui_3_16_0_1_1416396176084_18377" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div class="yiv8708658324y_msg_container" id="yiv8708658324yui_3_16_0_1_1416396176084_18376"> <hr id="yiv8708658324yui_3_16_0_1_1416396176084_18608" size="1"><br clear="none"><div dir="ltr" id="yiv8708658324yui_3_16_0_1_1416396176084_18536">Jssip (like <a rel="nofollow" shape="rect" target="_blank" href="http://tryit.jssip.net/">http://tryit.jssip.net/</a>) work with wss URLs too. Use the<br clear="none">ones I specified earlier in the thread.<br clear="none"><br clear="none">Andreas<br clear="none"><br clear="none">On 11/19/2014 12:32 PM, H Yavari wrote:<br clear="none">> Hi,<br clear="none">> <br clear="none">> Thanks. But my problem is that clients like jssip only support ws://<ip>.<br clear="none">> after upgrading to mr3.6.1, what address I should use for WS URI ????<br clear="none">> (WS URI is mandatory field)<br clear="none">> <br clear="none">> <br clear="none">> Regards,<br clear="none">> H. Yavari<br clear="none">> <br></div></div></div></div></div></div></div></div><br></div> </div> </div> </div></body></html>