<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="">And the first one is sipml5?<div class="">In my config both are working.</div><div class="">Check again your script. There is not one place, where automatic detection is done.</div><div class="">I don’t exactly remember. It was about 4-5 month ago. But I think, difference is between ws and wss.</div><div class="">Sorry, I can not publish my script. There are many other things, including push notifications.</div><div class="">I think Sipwise will be not happy, if I publish it.</div><div class=""><br class=""></div><div class="">Regards,</div><div class="">Nikita Stashkov<br class=""><div><blockquote type="cite" class=""><div class="">24 нояб. 2014 г., в 11:45, H Yavari <<a href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>> написал(а):</div><br class="Apple-interchange-newline"><div class=""><div class=""><div style="background-color: rgb(255, 255, 255); font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-size: 13px;" class=""><div dir="ltr" id="yui_3_16_0_1_1416808370215_10217" class=""><span class="">Hi,</span></div><div id="yui_3_16_0_1_1416808370215_10372" dir="ltr" class=""><span class=""><br class=""></span></div><div id="yui_3_16_0_1_1416808370215_10336" class="">I noticed a new thing that when I using jssip, calls not terminated. in the logs and I didn't see any rtcp-mux. so this two webRTC client is different in using SDP params?</div><div id="yui_3_16_0_1_1416808370215_10335" class=""><br class=""></div><div id="yui_3_16_0_1_1416808370215_10334" dir="ltr" class=""><br class="">Regards,</div><div id="yui_3_16_0_1_1416808370215_10361" dir="ltr" class="">H.Yavari<br class=""></div><div id="yui_3_16_0_1_1416808370215_10275" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;" class=""> <div id="yui_3_16_0_1_1416808370215_10274" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;" class=""> <div id="yui_3_16_0_1_1416808370215_10273" dir="ltr" class=""> <hr id="yui_3_16_0_1_1416808370215_10362" size="1" class=""> <font id="yui_3_16_0_1_1416808370215_10278" face="Arial" size="2" class=""> <b id="yui_3_16_0_1_1416808370215_10277" class=""><span id="yui_3_16_0_1_1416808370215_10276" style="font-weight:bold;" class="">From:</span></b> Nikita Stashkov <<a href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>><br class=""> <br class=""> </font> </div> <div id="yui_3_16_0_1_1416808370215_10297" class="y_msg_container"><br class=""><div id="yiv5295838733" class=""><div id="yui_3_16_0_1_1416808370215_10296" class=""><div id="yui_3_16_0_1_1416808370215_10302" class="">There may be different logic. I am doing it if caller is WebRTC, and callee is SIP. </div><div id="yui_3_16_0_1_1416808370215_10301" class="">Simply add this flag, when calling Rtpengine, like all other flags.</div><div id="yui_3_16_0_1_1416808370215_10300" class="">You can do nothing if both are WebRTC.</div><div id="yui_3_16_0_1_1416808370215_10363" class=""><br clear="none" class=""></div><div id="yui_3_16_0_1_1416808370215_10299" class="">Regards,</div><div id="yui_3_16_0_1_1416808370215_10298" class="">Nikita Stashkov<br clear="none" class=""><br clear="none" class=""><br clear="none" class=""></div><div id="yui_3_16_0_1_1416808370215_10295" class=""><br clear="none" class="">24 нояб. 2014 г., в 11:13, H Yavari <<a rel="nofollow" shape="rect" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>> написал(а):<br clear="none" class=""><br clear="none" class=""></div><div class="qtdSeparateBR"><br class=""><br class=""></div><div class="yiv5295838733yqt3694124101" id="yiv5295838733yqt24914"><blockquote type="cite" class=""><div class=""><div style="background-color: rgb(255, 255, 255); font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-size: 13px;" class=""><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2955" class=""><span class="">Hi,</span></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2987" class=""><span id="yiv5295838733yui_3_16_0_1_1416808370215_2986" class="">I checked it. the client (webRTC browser-sipml5) send rtp-mux. and is in the rtp.log too. so how can I disable this? or how can I add rtcp-mux-demux ? I should do this for all calls? or only for webRTC client?</span></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2985" class=""><br clear="none" class=""><span class=""></span></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2984" class=""><br clear="none" class=""><span class=""></span></div><div id="yiv5295838733yui_3_16_0_1_1416808370215_2983" class="">Thanks a lot.</div><div id="yiv5295838733yui_3_16_0_1_1416808370215_3143" class=""><br clear="none" class=""></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2982" class="">Regards,</div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2981" class="">H.Yavari</div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2979" class=""><br clear="none" class=""> </div><div id="yiv5295838733yui_3_16_0_1_1416808370215_2958" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;" class=""> <div id="yiv5295838733yui_3_16_0_1_1416808370215_2957" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;" class=""> <div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416808370215_2956" class=""> <hr id="yiv5295838733yui_3_16_0_1_1416808370215_2980" size="1" class=""> <font id="yiv5295838733yui_3_16_0_1_1416808370215_2959" face="Arial" size="2" class=""> <b class=""><span style="font-weight:bold;" class="">From:</span></b> Nikita Stashkov <<a rel="nofollow" shape="rect" ymailto="mailto:snl@sipmobile.org" target="_blank" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>><br clear="none" class=""> <b class=""><span style="font-weight:bold;" class=""></span></b><br clear="none" class=""> </font> </div> <div class="yiv5295838733y_msg_container" id="yiv5295838733yui_3_16_0_1_1416808370215_2969"><br clear="none" class=""><div id="yiv5295838733" class=""><div id="yiv5295838733yui_3_16_0_1_1416808370215_2968" class=""><div id="yiv5295838733yui_3_16_0_1_1416808370215_2978" class="">You need to look logs from WebRTC client and Pcap from SIP.</div><div id="yiv5295838733yui_3_16_0_1_1416808370215_2967" class="">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 clear="none" class=""><br clear="none" class="">Regards,</div><div id="yiv5295838733yui_3_16_0_1_1416808370215_2970" class="">Nikita Stashkov<br clear="none" class=""><br clear="none" class=""></div><div id="yiv5295838733yui_3_16_0_1_1416808370215_2977" class=""><br clear="none" class="">23. nov. 2014, в 13.40, H Yavari <<a rel="nofollow" shape="rect" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>> написал(а):<br clear="none" class=""><br clear="none" class=""></div><div class="yiv5295838733qtdSeparateBR"><br clear="none" class=""><br clear="none" class=""></div><div class="yiv5295838733yqt7534894036" id="yiv5295838733yqt01940"><blockquote id="yiv5295838733yui_3_16_0_1_1416808370215_2973" type="cite" class=""><div id="yiv5295838733yui_3_16_0_1_1416808370215_2972" class=""><div id="yiv5295838733yui_3_16_0_1_1416808370215_2971" style="background-color: rgb(255, 255, 255); font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-size: 13px;" class=""><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_7618" class=""><span class="">Hi,</span></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8691" class=""><span class=""><br clear="none" class=""></span></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8351" class=""><span id="yiv5295838733yui_3_16_0_1_1416740150963_8352" class="">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 clear="none" class=""></span></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8410" class=""><span id="yiv5295838733yui_3_16_0_1_1416740150963_8352" class="">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 dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8409" class="">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 dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8408" class=""><br clear="none" class=""></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8387" class="">So situation is complicated :)</div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8404" class="">SPCE specialist plz help!</div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8405" class=""><br clear="none" class=""></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8482" class=""><br clear="none" class=""></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8406" class="">Regards,</div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8407" class="">H. Yavari<br clear="none" class=""></div><div dir="ltr" id="yiv5295838733yui_3_16_0_1_1416740150963_8386" class=""><span id="yiv5295838733yui_3_16_0_1_1416740150963_8352" class=""></span></div><br clear="none" class=""> <div id="yiv5295838733yui_3_16_0_1_1416740150963_7556" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;" class=""> <div id="yiv5295838733yui_3_16_0_1_1416740150963_7555" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;" class=""> <div class="yiv5295838733y_msg_container" id="yiv5295838733yui_3_16_0_1_1416740150963_7560"> <hr id="yiv5295838733yui_3_16_0_1_1416740150963_8685" size="1" class=""> <font id="yiv5295838733yui_3_16_0_1_1416740150963_7557" face="Arial" size="2" class=""> <b id="yiv5295838733yui_3_16_0_1_1416740150963_8357" class=""><span id="yiv5295838733yui_3_16_0_1_1416740150963_8356" style="font-weight:bold;" class="">From:</span></b> Nikita Stashkov <<a rel="nofollow" shape="rect" ymailto="mailto:snl@sipmobile.org" target="_blank" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>><br clear="none" class=""><b id="yiv5295838733yui_3_16_0_1_1416740150963_7562" class=""><span id="yiv5295838733yui_3_16_0_1_1416740150963_7561" style="font-weight:bold;" class=""></span></b></font><br clear="none" class=""><div id="yiv5295838733" class=""><div id="yiv5295838733yui_3_16_0_1_1416740150963_7563" class="">Sorry, I can not share my script.<div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7564">What can you do.</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7565">Look the script /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.tt2</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7566">Of course, before modifying copy it to proxy.cfg.customtt.tt2</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7567">I think, webrtc endpoint automatic detection is not working for you.</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7568">It must look like this:</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7569"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7570">if($(ru{uri.param,transport}) == "ws" || $(ru{uri.param,transport}) == "wss»)</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7571"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7572">Then check flags you are sending to rtpengine.</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7573">To call SIP clients you must use flag rtcp-mux-demux</div><div class="yiv5295838733"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733">Regards,</div><div class="yiv5295838733">Nikita Stashkov</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7574"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7580"><br class="yiv5295838733" clear="none"><div id="yiv5295838733yui_3_16_0_1_1416740150963_7579" class=""><blockquote class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7578" type="cite"><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7581">22 нояб. 2014 г., в 20:28, H Yavari <<a rel="nofollow" shape="rect" class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7582" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>> написал(а):</div><br class="yiv5295838733Apple-interchange-newline" clear="none"><div class="yiv5295838733qtdSeparateBR"><br clear="none" class=""><br clear="none" class=""></div><div class="yiv5295838733yqt6899690072" id="yiv5295838733yqt54362"><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7577"><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7576"><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7575" style="background-color:rgb(255, 255, 255);font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-size:13px;"><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6780">Hi,</div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6780">I checked you site. it seems that is a good webRTC solution.</div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6780">Can you share with us your experience to solve our problem? or any script modifications?</div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6780"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6780"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6780">Regards,</div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6780">H.Yavari</div><br class="yiv5295838733" clear="none"> <div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6784" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6783" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6782"> <hr class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6781" size="1"> <font class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6787" face="Arial" size="2"> <b class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6786"><span class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6785" style="font-weight:bold;">From:</span></b> Nikita Stashkov <<a rel="nofollow" shape="rect" class="yiv5295838733" ymailto="mailto:snl@sipmobile.org" target="_blank" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>><br class="yiv5295838733" clear="none"></font></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416682321572_6782"><font class="yiv5295838733" face="Arial" size="2"><br class="yiv5295838733" clear="none"></font></div><div class="yiv5295838733y_msg_container" id="yiv5295838733yui_3_16_0_1_1416682321572_6788"><div class="yiv5295838733" id="yiv5295838733"><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6803">You can try with my site - <a rel="nofollow" shape="rect" class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6824" target="_blank" href="http://www.sipmobile.org/">www.sipmobile.org</a>.<div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6804">Create accounts: <a rel="nofollow" shape="rect" class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6825" target="_blank" href="https://www.sipmobile.org/register/">https://www.sipmobile.org/register/</a></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6826">And try to call with webRTC client and SIP.</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6934">I have modified some Kamailio SPCE scripts.</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6935"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6936">Regards,</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6937">Nikita Stashkov</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6938"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416682321572_6939"><br class="yiv5295838733" clear="none"><div class="yiv5295838733"><blockquote class="yiv5295838733" type="cite"><div class="yiv5295838733">22 нояб. 2014 г., в 16:04, Thomas Odorfer <<a rel="nofollow" shape="rect" class="yiv5295838733" ymailto="mailto:odotom@gmail.com" target="_blank" href="mailto:odotom@gmail.com">odotom@gmail.com</a>> написал(а):</div><br class="yiv5295838733Apple-interchange-newline" clear="none"><div class="yiv5295838733"></div></blockquote></div></div></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7585"><div class="yiv5295838733qtdSeparateBR"><br class="yiv5295838733" clear="none"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733yqt7138267133" id="yiv5295838733yqt88872"><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7584" style="word-wrap:break-word;">Hi,<div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7583">not sure if I understood correctly which scenario works and which not. </div><div class="yiv5295838733">So browser to soft phone is now working, but what is the meaning of browser to client? Which client?</div><div class="yiv5295838733"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733">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="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7602">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="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7603">(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="yiv5295838733">However, after my changes the following tests had been successful:</div><div class="yiv5295838733">browser webrtc to softphone (eg. jitsi, counterpath x-lite - should be software compatible with eyebeam)</div><div class="yiv5295838733">browser webrtc to another browser webrtc (jssip-0.50)</div><div class="yiv5295838733">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="yiv5295838733"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733">That is based on the latest SPCE version 3.6.1.</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7610">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="yiv5295838733">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="yiv5295838733"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733">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="yiv5295838733"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733">Good luck</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7611">Thomas</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7612"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_7613"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416740150963_8346">Am 22.11.2014 um 12:49 schrieb H Yavari <<a rel="nofollow" shape="rect" class="yiv5295838733" ymailto="mailto:hyavari@rocketmail.com" target="_blank" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>:<div class="yiv5295838733"><br class="yiv5295838733Apple-interchange-newline" clear="none"><blockquote class="yiv5295838733" type="cite"><div class="yiv5295838733" style="background-color:rgb(255, 255, 255);font-family:HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif;font-size:13px;"><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2631">Hi,</div><div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2650"><br class="yiv5295838733" clear="none"></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2630">I did this configs:</div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416469943651_153447" style=""><span class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416469943651_153428" style="">use_rtpproxy: „Always with rtpptoxy as only ICE candidate“ <br class="yiv5295838733" style="" clear="none"></span></div>rtcp_feedback: „Force AVP“ <br class="yiv5295838733" style="" clear="none">srtp_transcoding: „Force RTP“<div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2593"><br class="yiv5295838733" clear="none"><span class="yiv5295838733"></span></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2629"><span class="yiv5295838733" id="yiv5295838733yui_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="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2626"><span class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2627">and calls not be established. Have you any idea about this situation? <br class="yiv5295838733" clear="none"></span></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2634">Thanks for helps.<br class="yiv5295838733" clear="none"><span class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2627"></span></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2635"><br class="yiv5295838733" clear="none"><span class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2627"></span></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2636"><span class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2627">Regards,</span></div><div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2637"><span class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2627">H.Yavari</span></div> <div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2596" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px;"> <div class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2595" style="font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif;font-size:16px;"> <div class="yiv5295838733" dir="ltr" id="yiv5295838733yui_3_16_0_1_1416651599493_2594"> <hr class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2624" size="1"> <font class="yiv5295838733" id="yiv5295838733yui_3_16_0_1_1416651599493_2597" face="Arial" size="2"> <br class="yiv5295838733" clear="none"> </font> </div> </div></div></div></blockquote></div><br class="yiv5295838733" clear="none"></div></div></div>_______________________________________________<br class="yiv5295838733" clear="none">Spce-user mailing list<br class="yiv5295838733" clear="none"><a rel="nofollow" shape="rect" class="yiv5295838733" ymailto="mailto:Spce-user@lists.sipwise.com" target="_blank" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><br class="yiv5295838733" clear="none"><a rel="nofollow" shape="rect" class="yiv5295838733" target="_blank" href="https://lists.sipwise.com/listinfo/spce-user">https://lists.sipwise.com/listinfo/spce-user</a><br class="yiv5295838733" clear="none"><br class="yiv5295838733" clear="none"></div></div><br class="yiv5295838733" clear="none"><br class="yiv5295838733" clear="none"></div> </div> </div> </div></div></div></div></blockquote></div><br class="yiv5295838733" clear="none"></div></div></div><br clear="none" class=""><br clear="none" class=""></div> </div> </div> </div></div></blockquote></div></div></div><br clear="none" class=""><br clear="none" class=""></div> </div> </div> </div></div></blockquote></div></div></div><br class=""><br class=""></div> </div> </div> </div></div></div></blockquote></div><br class=""></div></body></html>