<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">You can try with my site - <a href="http://www.sipmobile.org" class="">www.sipmobile.org</a>.<div class="">Create accounts: <a href="https://www.sipmobile.org/register/" class="">https://www.sipmobile.org/register/</a></div><div class="">And try to call with webRTC client and SIP.</div><div class="">I have modified some Kamailio SPCE scripts.</div><div class=""><br class=""></div><div class="">Regards,</div><div class="">Nikita Stashkov</div><div class=""><br class=""></div><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">22 нояб. 2014 г., в 16:04, Thomas Odorfer <<a href="mailto:odotom@gmail.com" class="">odotom@gmail.com</a>> написал(а):</div><br class="Apple-interchange-newline"><div class=""><meta http-equiv="Content-Type" content="text/html charset=windows-1252" class=""><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hi,<div class="">not sure if I understood correctly which scenario works and which not. </div><div class="">So browser to soft phone is now working, but what is the meaning of browser to client? Which client?</div><div class=""><br class=""></div><div class="">I tested myself and I have to confess that I had to do some changes in the account configs for soft phones where I am not happy about.</div><div class="">It only worked between browser-webrtc and soft phone when the corresponding account for the soft phone - nat & media flow control had been changed to "force avp"“ and "force rtp“ ie. no encryption.</div><div class="">(I have to investigate that one - could be related to an upgrade I had performed last week - usually srtp should also work with softphones, within the ftp.log there was „SRTP output wanted but no crypto suite was negotiated“).</div><div class="">However, after my changes the following tests had been successful:</div><div class="">browser webrtc to softphone (eg. jitsi, counterpath x-lite - should be software compatible with eyebeam)</div><div class="">browser webrtc to another browser webrtc (jssip-0.50)</div><div class="">browser webrtc to pstn via sip trunking (standard sip trunk, peer settings for media force „rtp“, „force rtp“, „always with plain SDP“)</div><div class=""><br class=""></div><div class="">That is based on the latest SPCE version 3.6.1.</div><div class="">What does not seem to be achievable at the moment that you can have an account that supports „standard“ and webrtc simultaneously ( at least I haven’t succeeded with such a setup, maybe some sipwise/kamailio/rtpengine expert knows the trick). And I do not have a solution yet how to share one phone number between two accounts with different profiles.</div><div class="">The only solution I have at the moment is that I put a webrtc gateway (similar to webrtc2sip from doubango) in front of SPCE for webrtc clients.</div><div class=""><br class=""></div><div class="">For your particular problem, maybe you have to check whether your domain settings allow „bypass rtp proxy“ behind the same NAT - assuming you are testing wthin your LAN - this should be set to never.</div><div class=""><br class=""></div><div class="">Good luck</div><div class="">Thomas</div><div class=""><br class=""></div><div class=""><br class=""></div><div class="">Am 22.11.2014 um 12:49 schrieb H Yavari <<a href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>:<div class=""><br class="Apple-interchange-newline"><blockquote type="cite" class=""><div style="background-color: rgb(255, 255, 255); font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-size: 13px; position: static; z-index: auto;" class=""><div id="yui_3_16_0_1_1416651599493_2631" class="">Hi,</div><div id="yui_3_16_0_1_1416651599493_2650" class=""><br class=""></div><div id="yui_3_16_0_1_1416651599493_2630" dir="ltr" class="">I did this configs:</div><div style="" class="" dir="ltr" id="yiv0853063581yui_3_16_0_1_1416469943651_153447"><span style="" class="" id="yiv0853063581yui_3_16_0_1_1416469943651_153428">use_rtpproxy: „Always with rtpptoxy as only ICE candidate“ <br style="" class="" clear="none"></span></div>rtcp_feedback: „Force AVP“ <br style="" class="" clear="none">srtp_transcoding: „Force RTP“<div id="yui_3_16_0_1_1416651599493_2593" class=""><br class=""><span class=""></span></div><div id="yui_3_16_0_1_1416651599493_2629" dir="ltr" class=""><span id="yui_3_16_0_1_1416651599493_2628" class="">now calls between browser to soft phone is ok, but browser to client and browser to browser receive this error "Failed to get local SDP"</span></div><div id="yui_3_16_0_1_1416651599493_2626" dir="ltr" class=""><span id="yui_3_16_0_1_1416651599493_2627" class="">and calls not be established. Have you any idea about this situation? <br class=""></span></div><div id="yui_3_16_0_1_1416651599493_2634" dir="ltr" class="">Thanks for helps.<br class=""><span id="yui_3_16_0_1_1416651599493_2627" class=""></span></div><div id="yui_3_16_0_1_1416651599493_2635" dir="ltr" class=""><br class=""><span id="yui_3_16_0_1_1416651599493_2627" class=""></span></div><div id="yui_3_16_0_1_1416651599493_2636" dir="ltr" class=""><span id="yui_3_16_0_1_1416651599493_2627" class="">Regards,</span></div><div id="yui_3_16_0_1_1416651599493_2637" dir="ltr" class=""><span id="yui_3_16_0_1_1416651599493_2627" class="">H.Yavari</span></div> <div id="yui_3_16_0_1_1416651599493_2596" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;" class=""> <div id="yui_3_16_0_1_1416651599493_2595" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;" class=""> <div id="yui_3_16_0_1_1416651599493_2594" dir="ltr" class=""> <hr id="yui_3_16_0_1_1416651599493_2624" size="1" class=""> <font id="yui_3_16_0_1_1416651599493_2597" face="Arial" size="2" class=""> <br class=""> </font> </div> </div></div></div></blockquote></div><br class=""></div></div>_______________________________________________<br class="">Spce-user mailing list<br class=""><a href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a><br class="">https://lists.sipwise.com/listinfo/spce-user<br class=""></div></blockquote></div><br class=""></div></body></html>