<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>You need to look logs from WebRTC client and Pcap from SIP.</div><div>Of cource, if SIP client recives SDP with rtp-mux, he will not understand it. And after 30 sec call will be terminated. But you must see logs. My system is based on SPCE 3.2, and manually compiled rtpengine. And I don't know was changed in current version. Also, you can look rtp.log. Sometimes it helps.<br><br>Regards,</div><div>Nikita Stashkov<br><br></div><div><br>23. nov. 2014, в 13.40, H Yavari <<a href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>> написал(а):<br><br></div><blockquote type="cite"><div><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px"><div dir="ltr" id="yui_3_16_0_1_1416740150963_7618"><span>Hi,</span></div><div id="yui_3_16_0_1_1416740150963_8691" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1416740150963_8351" dir="ltr"><span id="yui_3_16_0_1_1416740150963_8352">Dear I did this before that I changed "ws" with "wss" but now after your reply I did "ws" || "wss". but not any changes.<br></span></div><div id="yui_3_16_0_1_1416740150963_8410" dir="ltr"><span id="yui_3_16_0_1_1416740150963_8352">As I told before, now my main problem is calls hangup after 30 sec. In your opinion the rtcp-mux-demux flags adding will solve this?</span></div><div id="yui_3_16_0_1_1416740150963_8409" dir="ltr">another point is that before 30 sec, If any call parties (caller: browser and callee: soft phone) hangs up, the call not terminate until 30 sec timeout. I think that the dialog of a call not recognized.</div><div id="yui_3_16_0_1_1416740150963_8408" dir="ltr"><br></div><div id="yui_3_16_0_1_1416740150963_8387" dir="ltr">So situation is complicated :)</div><div id="yui_3_16_0_1_1416740150963_8404" dir="ltr">SPCE specialist plz help!</div><div id="yui_3_16_0_1_1416740150963_8405" dir="ltr"><br></div><div id="yui_3_16_0_1_1416740150963_8482" dir="ltr"><br></div><div id="yui_3_16_0_1_1416740150963_8406" dir="ltr">Regards,</div><div id="yui_3_16_0_1_1416740150963_8407" dir="ltr">H. Yavari<br></div><div id="yui_3_16_0_1_1416740150963_8386" dir="ltr"><span id="yui_3_16_0_1_1416740150963_8352"></span></div><br> <div id="yui_3_16_0_1_1416740150963_7556" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;"> <div id="yui_3_16_0_1_1416740150963_7555" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1416740150963_7560" class="y_msg_container"> <hr id="yui_3_16_0_1_1416740150963_8685" size="1"> <font id="yui_3_16_0_1_1416740150963_7557" face="Arial" size="2"> <b id="yui_3_16_0_1_1416740150963_8357"><span id="yui_3_16_0_1_1416740150963_8356" style="font-weight:bold;">From:</span></b> Nikita Stashkov <<a href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>><br><b id="yui_3_16_0_1_1416740150963_7562"><span id="yui_3_16_0_1_1416740150963_7561" style="font-weight: bold;"></span></b></font><br><div id="yiv9192471702"><div id="yui_3_16_0_1_1416740150963_7563">Sorry, I can not share my script.<div id="yui_3_16_0_1_1416740150963_7564" class="yiv9192471702">What can you do.</div><div id="yui_3_16_0_1_1416740150963_7565" class="yiv9192471702">Look the script /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.tt2</div><div id="yui_3_16_0_1_1416740150963_7566" class="yiv9192471702">Of course, before modifying copy it to proxy.cfg.customtt.tt2</div><div id="yui_3_16_0_1_1416740150963_7567" class="yiv9192471702">I think, webrtc endpoint automatic detection is not working for you.</div><div id="yui_3_16_0_1_1416740150963_7568" class="yiv9192471702">It must look like this:</div><div id="yui_3_16_0_1_1416740150963_7569" class="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div id="yui_3_16_0_1_1416740150963_7570" class="yiv9192471702">if($(ru{uri.param,transport}) == "ws" || $(ru{uri.param,transport}) == "wss»)</div><div id="yui_3_16_0_1_1416740150963_7571" class="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div id="yui_3_16_0_1_1416740150963_7572" class="yiv9192471702">Then check flags you are sending to rtpengine.</div><div id="yui_3_16_0_1_1416740150963_7573" class="yiv9192471702">To call SIP clients you must use flag rtcp-mux-demux</div><div class="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702">Regards,</div><div class="yiv9192471702">Nikita Stashkov</div><div id="yui_3_16_0_1_1416740150963_7574" class="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div id="yui_3_16_0_1_1416740150963_7580" class="yiv9192471702"><br class="yiv9192471702" clear="none"><div id="yui_3_16_0_1_1416740150963_7579"><blockquote id="yui_3_16_0_1_1416740150963_7578" class="yiv9192471702" type="cite"><div id="yui_3_16_0_1_1416740150963_7581" class="yiv9192471702">22 нояб. 2014 г., в 20:28, H Yavari <<a id="yui_3_16_0_1_1416740150963_7582" rel="nofollow" shape="rect" class="yiv9192471702" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>> написал(а):</div><br class="yiv9192471702Apple-interchange-newline" clear="none"><div class="qtdSeparateBR"><br><br></div><div class="yiv9192471702yqt6899690072" id="yiv9192471702yqt54362"><div id="yui_3_16_0_1_1416740150963_7577" class="yiv9192471702"><div id="yui_3_16_0_1_1416740150963_7576" class="yiv9192471702"><div id="yui_3_16_0_1_1416740150963_7575" class="yiv9192471702" style="background-color:rgb(255, 255, 255);font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-size:13px;"><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6780">Hi,</div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6780">I checked you site. it seems that is a good webRTC solution.</div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6780">Can you share with us your experience to solve our problem? or any script modifications?</div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6780"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6780"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6780">Regards,</div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6780">H.Yavari</div><br class="yiv9192471702" clear="none"> <div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6784" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6783" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6782"> <hr class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6781" size="1"> <font class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6787" face="Arial" size="2"> <b class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6786"><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6785" style="font-weight:bold;">From:</span></b> Nikita Stashkov <<a rel="nofollow" shape="rect" class="yiv9192471702" ymailto="mailto:snl@sipmobile.org" target="_blank" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>><br class="yiv9192471702" clear="none"></font></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416682321572_6782"><font class="yiv9192471702" face="Arial" size="2"><br class="yiv9192471702" clear="none"></font></div><div class="yiv9192471702y_msg_container" id="yiv9192471702yui_3_16_0_1_1416682321572_6788"><div class="yiv9192471702" id="yiv9192471702"><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6803">You can try with my site - <a rel="nofollow" shape="rect" class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6824" target="_blank" href="http://www.sipmobile.org/">www.sipmobile.org</a>.<div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6804">Create accounts: <a rel="nofollow" shape="rect" class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6825" target="_blank" href="https://www.sipmobile.org/register/">https://www.sipmobile.org/register/</a></div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6826">And try to call with webRTC client and SIP.</div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6934">I have modified some Kamailio SPCE scripts.</div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6935"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6936">Regards,</div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6937">Nikita Stashkov</div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6938"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416682321572_6939"><br class="yiv9192471702" clear="none"><div class="yiv9192471702"><blockquote class="yiv9192471702" type="cite"><div class="yiv9192471702">22 нояб. 2014 г., в 16:04, Thomas Odorfer <<a rel="nofollow" shape="rect" class="yiv9192471702" ymailto="mailto:odotom@gmail.com" target="_blank" href="mailto:odotom@gmail.com">odotom@gmail.com</a>> написал(а):</div><br class="yiv9192471702Apple-interchange-newline" clear="none"><div class="yiv9192471702"></div></blockquote></div></div></div><div id="yui_3_16_0_1_1416740150963_7585" class="yiv9192471702"><div class="yiv9192471702qtdSeparateBR"><br class="yiv9192471702" clear="none"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702yqt7138267133" id="yiv9192471702yqt88872"><div id="yui_3_16_0_1_1416740150963_7584" class="yiv9192471702" style="word-wrap:break-word;">Hi,<div id="yui_3_16_0_1_1416740150963_7583" class="yiv9192471702">not sure if I understood correctly which scenario works and which not. </div><div class="yiv9192471702">So browser to soft phone is now working, but what is the meaning of browser to client? Which client?</div><div class="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702">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 id="yui_3_16_0_1_1416740150963_7602" class="yiv9192471702">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 id="yui_3_16_0_1_1416740150963_7603" class="yiv9192471702">(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="yiv9192471702">However, after my changes the following tests had been successful:</div><div class="yiv9192471702">browser webrtc to softphone (eg. jitsi, counterpath x-lite - should be software compatible with eyebeam)</div><div class="yiv9192471702">browser webrtc to another browser webrtc (jssip-0.50)</div><div class="yiv9192471702">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="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702">That is based on the latest SPCE version 3.6.1.</div><div id="yui_3_16_0_1_1416740150963_7610" class="yiv9192471702">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="yiv9192471702">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="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702">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="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702">Good luck</div><div id="yui_3_16_0_1_1416740150963_7611" class="yiv9192471702">Thomas</div><div id="yui_3_16_0_1_1416740150963_7612" class="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div id="yui_3_16_0_1_1416740150963_7613" class="yiv9192471702"><br class="yiv9192471702" clear="none"></div><div id="yui_3_16_0_1_1416740150963_8346" class="yiv9192471702">Am 22.11.2014 um 12:49 schrieb H Yavari <<a rel="nofollow" shape="rect" class="yiv9192471702" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>:<div class="yiv9192471702"><br class="yiv9192471702Apple-interchange-newline" clear="none"><blockquote class="yiv9192471702" type="cite"><div class="yiv9192471702" style="background-color:rgb(255, 255, 255);font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-size:13px;"><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2631">Hi,</div><div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2650"><br class="yiv9192471702" clear="none"></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2630">I did this configs:</div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416469943651_153447" style=""><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416469943651_153428" style="">use_rtpproxy: „Always with rtpptoxy as only ICE candidate“ <br class="yiv9192471702" style="" clear="none"></span></div>rtcp_feedback: „Force AVP“ <br class="yiv9192471702" style="" clear="none">srtp_transcoding: „Force RTP“<div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2593"><br class="yiv9192471702" clear="none"><span class="yiv9192471702"></span></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2629"><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2628">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 class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2626"><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2627">and calls not be established. Have you any idea about this situation? <br class="yiv9192471702" clear="none"></span></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2634">Thanks for helps.<br class="yiv9192471702" clear="none"><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2627"></span></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2635"><br class="yiv9192471702" clear="none"><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2627"></span></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2636"><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2627">Regards,</span></div><div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2637"><span class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2627">H.Yavari</span></div> <div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2596" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2595" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div class="yiv9192471702" dir="ltr" id="yiv9192471702yui_3_16_0_1_1416651599493_2594"> <hr class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2624" size="1"> <font class="yiv9192471702" id="yiv9192471702yui_3_16_0_1_1416651599493_2597" face="Arial" size="2"> <br class="yiv9192471702" clear="none"> </font> </div> </div></div></div></blockquote></div><br class="yiv9192471702" clear="none"></div></div></div>_______________________________________________<br class="yiv9192471702" clear="none">Spce-user mailing list<br class="yiv9192471702" clear="none"><a rel="nofollow" shape="rect" class="yiv9192471702" ymailto="mailto:Spce-user@lists.sipwise.com" target="_blank" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><br class="yiv9192471702" clear="none"><a rel="nofollow" shape="rect" class="yiv9192471702" target="_blank" href="https://lists.sipwise.com/listinfo/spce-user">https://lists.sipwise.com/listinfo/spce-user</a><br class="yiv9192471702" clear="none"><br class="yiv9192471702" clear="none"></div></div><br class="yiv9192471702" clear="none"><br class="yiv9192471702" clear="none"></div> </div> </div> </div></div></div></div></blockquote></div><br class="yiv9192471702" clear="none"></div></div></div><br><br></div> </div> </div> </div></div></blockquote></body></html>