<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;">That is strange - I will try to test that setup myself (probably not with eyebeam but with xlite - from the same company, but for free)<div><br></div><div><br><div><div>Am 22.11.2014 um 07:16 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_153348"><span>Hi,</span></div><div id="yui_3_16_0_1_1416469943651_153430" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1416469943651_153427" dir="ltr"><span id="yui_3_16_0_1_1416469943651_153428">I did this configs,</span></div><div id="yui_3_16_0_1_1416469943651_153447" dir="ltr"><span id="yui_3_16_0_1_1416469943651_153428">use_rtpproxy: „Always with rtpptoxy as only ICE candidate“ <br></span></div><div id="yui_3_16_0_1_1416469943651_153441" dir="ltr">rtcp_feedback: „Prefer AVPF“ <br>srtp_transcoding: „Prefer SRTP“ <br style="" class=""></div><div id="yui_3_16_0_1_1416469943651_153440" dir="ltr"><span id="yui_3_16_0_1_1416469943651_153428"><br></span></div><div id="yui_3_16_0_1_1416469943651_153439" dir="ltr"><span id="yui_3_16_0_1_1416469943651_153428"> but the result was not good. the call was being in the "Call Progress" until timeout and called party (soft phone: eyebeam) not went in the ringing state anymore.</span></div><div id="yui_3_16_0_1_1416469943651_153426" dir="ltr">I attached the RTP logs too. plz help me to solve this issue.</div><div id="yui_3_16_0_1_1416469943651_153425" dir="ltr"><br></div><div id="yui_3_16_0_1_1416469943651_153424" dir="ltr"><br></div><div id="yui_3_16_0_1_1416469943651_153423" dir="ltr">Regards,</div><div id="yui_3_16_0_1_1416469943651_153422" dir="ltr">H.Yavari<br><span></span></div><div id="yui_3_16_0_1_1416469943651_153421" dir="ltr"><span></span></div><br> <div id="yui_3_16_0_1_1416469943651_153351" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;"> <div id="yui_3_16_0_1_1416469943651_153350" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1416469943651_153353" class="y_msg_container"> <hr id="yui_3_16_0_1_1416469943651_153407" size="1">Actually, I am using the following settings<div id="yiv6375969298"><div id="yui_3_16_0_1_1416469943651_153354"><div id="yui_3_16_0_1_1416469943651_153358"><br clear="none"></div><div id="yui_3_16_0_1_1416469943651_153359">In order to achieve transcoding/media adaptation with rtpengine, include rtpproxy for calling other VOIP phones/PSTN:</div><div id="yui_3_16_0_1_1416469943651_153360">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 id="yui_3_16_0_1_1416469943651_153361"><br clear="none"></div><div id="yui_3_16_0_1_1416469943651_153362">srtp_transcoding: „Prefer SRTP“ (webrtc mandates DTLS SRTP as media protocol, your original setting Force RTP prevents this)</div><div id="yui_3_16_0_1_1416469943651_153363">rtcp_feedback: „Prefer AVPF“ (AVPF is webrtc standard for audio/video profile control)</div><div id="yui_3_16_0_1_1416469943651_153364"><br clear="none"></div><div id="yui_3_16_0_1_1416469943651_153385">Maybe you try that with jssip first.</div><div id="yui_3_16_0_1_1416469943651_153384"><br clear="none"></div><div id="yui_3_16_0_1_1416469943651_153365"><br clear="none"></div><div id="yui_3_16_0_1_1416469943651_153386"><br clear="none"></div><div id="yui_3_16_0_1_1416469943651_153369"><br clear="none"><div id="yui_3_16_0_1_1416469943651_153368"><div id="yui_3_16_0_1_1416469943651_153387">Am 20.11.2014 um 10:29 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="yiv6375969298Apple-interchange-newline" clear="none"><div class="qtdSeparateBR"><br><br></div><div class="yiv6375969298yqt3036487232" id="yiv6375969298yqt14461"><blockquote id="yui_3_16_0_1_1416469943651_153367" type="cite"><div id="yui_3_16_0_1_1416469943651_153366" 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="yiv6375969298yui_3_16_0_1_1416469943651_12937"><span id="yiv6375969298yui_3_16_0_1_1416469943651_13107">Hi Thomas,</span></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13064"><br clear="none"><span></span></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13063"><span class="yiv6375969298" id="yiv6375969298yui_3_16_0_1_1416469943651_13108" style="">I using : "</span><span class="yiv6375969298" id="yiv6375969298yui_3_16_0_1_1416469943651_13086" style="">srtp_transcoding" = "Force RTP" and "</span><span id="yiv6375969298yui_3_16_0_1_1416469943651_13086">rtcp_feedback" = "Force AVP" other things are default of sipwise.</span></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13106">Thanks for reply.</div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13105"><br clear="none"></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13104"><br clear="none"><span id="yiv6375969298yui_3_16_0_1_1416469943651_13086"></span></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13098"><span id="yiv6375969298yui_3_16_0_1_1416469943651_13086">Regards,</span></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13103"><span id="yiv6375969298yui_3_16_0_1_1416469943651_13086">H. Yavari</span></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_13099"><span id="yiv6375969298yui_3_16_0_1_1416469943651_13086"><br clear="none"></span></div> <div id="yiv6375969298yui_3_16_0_1_1416469943651_12880" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div id="yiv6375969298yui_3_16_0_1_1416469943651_12879" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div class="yiv6375969298y_msg_container" id="yiv6375969298yui_3_16_0_1_1416469943651_13041"> <hr id="yiv6375969298yui_3_16_0_1_1416469943651_12877" size="1"><font id="yiv6375969298yui_3_16_0_1_1416469943651_12914" face="Arial" size="2"><b><span style="font-weight:bold;"></span></b></font>Hello Yavari,<div id="yiv6375969298"><div id="yiv6375969298yui_3_16_0_1_1416469943651_13042"><div id="yiv6375969298yui_3_16_0_1_1416469943651_13061">what are your current settings in „NAT and Media Flow Control“ for your sip user ?</div><div id="yiv6375969298yui_3_16_0_1_1416469943651_13060">jssip should actually work.</div><div id="yiv6375969298yui_3_16_0_1_1416469943651_13059">BR</div><div id="yiv6375969298yui_3_16_0_1_1416469943651_13058">Thomas</div><div id="yiv6375969298yui_3_16_0_1_1416469943651_13049"><br clear="none"><div id="yiv6375969298yui_3_16_0_1_1416469943651_13051"><div id="yiv6375969298yui_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="yiv6375969298Apple-interchange-newline" clear="none"><blockquote id="yiv6375969298yui_3_16_0_1_1416469943651_13054" type="cite"><div id="yui_3_16_0_1_1416469943651_153373" class="yiv6375969298qtdSeparateBR"><br clear="none"><br clear="none"></div><div class="yiv6375969298yqt5717883244" id="yiv6375969298yqt81894"><div id="yiv6375969298yui_3_16_0_1_1416469943651_13053"><div id="yiv6375969298yui_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="yiv6375969298yui_3_16_0_1_1416469943651_2403">Hi,</div><div dir="ltr" id="yiv6375969298yui_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="yiv6375969298yui_3_16_0_1_1416469943651_2461">So are there any configs that I should do ?</div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_2462">Thanks.</div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_2463"><br clear="none"></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_2464"><br clear="none"></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_2465"><br clear="none"></div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_2466">Regards,</div><div dir="ltr" id="yiv6375969298yui_3_16_0_1_1416469943651_2467">H.Yavari<br clear="none"></div><div id="yiv6375969298yui_3_16_0_1_1416469943651_2400"><span></span></div><br clear="none"> <div id="yiv6375969298yui_3_16_0_1_1416469943651_2457" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div id="yiv6375969298yui_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="yiv6375969298yui_3_16_0_1_1416469943651_2458"> <hr id="yui_3_16_0_1_1416469943651_153372" size="1"><font id="yiv6375969298yui_3_16_0_1_1416469943651_2459" face="Arial" size="2"><b><span style="font-weight:bold;"></span></b></font><br></div></div></div></div></div></div></blockquote></div></div></div></div><br></div> </div> </div> </div></blockquote></div></div><br clear="none"></div></div></div><br><br></div> </div> </div> </div></blockquote></div><br></div></body></html>