<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">Actually, I am using the following settings<div><br></div><div>In order to achieve transcoding/media adaptation with rtpengine, include rtpproxy for calling other VOIP phones/PSTN:</div><div>use_rtpproxy: „Always with rtpptoxy as only ICE candidate“ (alternatively „as additional ICE candidate“ - this might however lead to problems when calling other softphones)</div><div><br></div><div>srtp_transcoding: „Prefer SRTP“ (webrtc mandates DTLS SRTP as media protocol, your original setting Force RTP prevents this)</div><div>rtcp_feedback: „Prefer AVPF“ (AVPF is webrtc standard for audio/video profile control)</div><div><br></div><div>Maybe you try that with jssip first.</div><div><br></div><div><br></div><div><br></div><div><br><div><div>Am 20.11.2014 um 10:29 schrieb H Yavari <<a href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="background-color: rgb(255, 255, 255); font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-size: 13px;"><div dir="ltr" id="yui_3_16_0_1_1416469943651_12937"><span id="yui_3_16_0_1_1416469943651_13107">Hi Thomas,</span></div><div id="yui_3_16_0_1_1416469943651_13064" dir="ltr"><br><span></span></div><div id="yui_3_16_0_1_1416469943651_13063" dir="ltr"><span id="yui_3_16_0_1_1416469943651_13108" style="" class="">I using : "</span><span style="" class="" id="yui_3_16_0_1_1416469943651_13086">srtp_transcoding" = "Force RTP" and "</span><span id="yui_3_16_0_1_1416469943651_13086">rtcp_feedback" = "Force AVP" other things are default of sipwise.</span></div><div id="yui_3_16_0_1_1416469943651_13106" dir="ltr">Thanks for reply.</div><div id="yui_3_16_0_1_1416469943651_13105" dir="ltr"><br></div><div id="yui_3_16_0_1_1416469943651_13104" dir="ltr"><br><span id="yui_3_16_0_1_1416469943651_13086"></span></div><div id="yui_3_16_0_1_1416469943651_13098" dir="ltr"><span id="yui_3_16_0_1_1416469943651_13086">Regards,</span></div><div id="yui_3_16_0_1_1416469943651_13103" dir="ltr"><span id="yui_3_16_0_1_1416469943651_13086">H. Yavari</span></div><div id="yui_3_16_0_1_1416469943651_13099" dir="ltr"><span id="yui_3_16_0_1_1416469943651_13086"><br></span></div> <div id="yui_3_16_0_1_1416469943651_12880" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;"> <div id="yui_3_16_0_1_1416469943651_12879" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1416469943651_13041" class="y_msg_container"> <hr id="yui_3_16_0_1_1416469943651_12877" size="1"><font id="yui_3_16_0_1_1416469943651_12914" face="Arial" size="2"><b><span style="font-weight:bold;"></span></b></font>Hello Yavari,<div id="yiv4544285680"><div id="yui_3_16_0_1_1416469943651_13042"><div id="yui_3_16_0_1_1416469943651_13061">what are your current settings in „NAT and Media Flow Control“ for your sip user ?</div><div id="yui_3_16_0_1_1416469943651_13060">jssip should actually work.</div><div id="yui_3_16_0_1_1416469943651_13059">BR</div><div id="yui_3_16_0_1_1416469943651_13058">Thomas</div><div id="yui_3_16_0_1_1416469943651_13049"><br clear="none"><div id="yui_3_16_0_1_1416469943651_13051"><div id="yui_3_16_0_1_1416469943651_13050">Am 20.11.2014 um 09:35 schrieb H Yavari <<a rel="nofollow" shape="rect" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>:</div><br class="yiv4544285680Apple-interchange-newline" clear="none"><blockquote id="yui_3_16_0_1_1416469943651_13054" type="cite"><div class="qtdSeparateBR"><br><br></div><div class="yiv4544285680yqt5717883244" id="yiv4544285680yqt81894"><div id="yui_3_16_0_1_1416469943651_13053"><div id="yui_3_16_0_1_1416469943651_13052" style="background-color:rgb(255, 255, 255);font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-size:13px;"><div id="yiv4544285680yui_3_16_0_1_1416469943651_2403">Hi,</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2404">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 dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2461">So are there any configs that I should do ?</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2462">Thanks.</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2463"><br clear="none"></div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2464"><br clear="none"></div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2465"><br clear="none"></div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2466">Regards,</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2467">H.Yavari<br clear="none"></div><div id="yiv4544285680yui_3_16_0_1_1416469943651_2400"><span></span></div><br clear="none"> <div id="yiv4544285680yui_3_16_0_1_1416469943651_2457" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div id="yiv4544285680yui_3_16_0_1_1416469943651_2456" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416469943651_2458"> <hr size="1"> <font id="yiv4544285680yui_3_16_0_1_1416469943651_2459" face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> H Yavari <<a rel="nofollow" shape="rect" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br clear="none"> <b id="yiv4544285680yui_3_16_0_1_1416469943651_2634"><span id="yiv4544285680yui_3_16_0_1_1416469943651_2633" style="font-weight:bold;">To:</span></b> "<a rel="nofollow" shape="rect" ymailto="mailto:spce-user@lists.sipwise.com" target="_blank" href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a>" <<a rel="nofollow" shape="rect" ymailto="mailto:spce-user@lists.sipwise.com" target="_blank" href="mailto:spce-user@lists.sipwise.com">spce-user@lists.sipwise.com</a>> <br clear="none"> <b id="yiv4544285680yui_3_16_0_1_1416469943651_2636"><span id="yiv4544285680yui_3_16_0_1_1416469943651_2635" style="font-weight:bold;">Sent:</span></b> Thursday, 20 November 2014, 10:12:08<br clear="none"> <b id="yiv4544285680yui_3_16_0_1_1416469943651_2632"><span id="yiv4544285680yui_3_16_0_1_1416469943651_2631" style="font-weight:bold;">Subject:</span></b> Re: [Spce-user] webRTC in production<br clear="none"> </font> </div> <div class="yiv4544285680y_msg_container" id="yiv4544285680yui_3_16_0_1_1416469943651_2455"><br clear="none"><div id="yiv4544285680"><div id="yiv4544285680yui_3_16_0_1_1416469943651_2454"><div id="yiv4544285680yui_3_16_0_1_1416469943651_2453" style="background-color:rgb(255, 255, 255);font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-size:13px;"><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18533"><span>Hi,</span></div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18582"><span id="yiv4544285680yui_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="yiv4544285680yui_3_16_0_1_1416396176084_18590">and call not established.</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18593"><br clear="none"></div><div dir="ltr" id="yiv4544285680yui_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="yiv4544285680yui_3_16_0_1_1416396176084_18622">How can I solve this issue?</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18612">Thanks.</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18613"><br clear="none"></div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18615"><br clear="none"></div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18617">Regards,</div><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18619">H.Yavari<div class="yiv4544285680qtdSeparateBR"><br clear="none"><br clear="none"></div><div class="yiv4544285680yqt1458283842" id="yiv4544285680yqtfd94043"><br clear="none"></div></div><div class="yiv4544285680yqt1458283842" id="yiv4544285680yqtfd79698"><div id="yiv4544285680yui_3_16_0_1_1416396176084_18378" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div id="yiv4544285680yui_3_16_0_1_1416396176084_18377" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div class="yiv4544285680y_msg_container" id="yiv4544285680yui_3_16_0_1_1416396176084_18376"> <hr id="yiv4544285680yui_3_16_0_1_1416396176084_18608" size="1"><br clear="none"><div dir="ltr" id="yiv4544285680yui_3_16_0_1_1416396176084_18536">Jssip (like <a id="yui_3_16_0_1_1416469943651_13057" 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"></div></div></div></div></div></div></div></div></div></div></div></div></div></div></blockquote></div><br></div></div></div><br><br></div> </div> </div> </div></blockquote></div><br></div></body></html>