<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Freeswitch already supports webrtc, there is no need for a gateway.<div class=""><br class=""></div><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Aug 23, 2016, at 5:00 AM, james <<a href="mailto:freepbx@qq.com" class="">freepbx@qq.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div class="">Hello:</div><div class="">I am using portsip webRTC gateway to test the compatibility. I use two phones for webRTC testing. One is normal webRTC client and another one is softphone.</div><div class="">FreeSWITCH work as a IPPBX. <span style="line-height: 1.5;" class="">The call flows are:</span></div><div class=""><span style="color: rgb(204, 0, 0); font-family: Arial, Helvetica; font-size: 13px; line-height: 18px;" class=""></span></div><div class=""><span id="cid:290CF6FD@45AAC238.9810BC57"><290CF6FD@45AAC238.9810BC57></span> </div><div class="">the problem is that webRTC calls sip phone without any problem, but sip phone calls to webRTC than failed. </div><div class="">errors are:</div><div class=""><div class="">2016-08-23 16:29:39.971775 [DEBUG] switch_core_state_machine.c:569 (<a href="mailto:sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid" class="">sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid</a>) State Change CS_REPORTING -> CS_DESTROY</div><div class="">freeswitch@iZ23lkvsnwpZ> </div><div class="">2016-08-23 16:29:39.971775 [DEBUG] switch_core_session.c:1647 Session 171 (<a href="mailto:sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid" class="">sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid</a>) Locked, Waiting on external entities</div><div class="">freeswitch@iZ23lkvsnwpZ> </div><div class="">2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]</div><div class="">freeswitch@iZ23lkvsnwpZ> </div><div class="">2016-08-23 16:29:39.981772 [NOTICE] switch_ivr_originate.c:2771 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]</div><div class="">freeswitch@iZ23lkvsnwpZ> </div><div class="">2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]</div><div class="">freeswitch@iZ23lkvsnwpZ> </div><div class="">2016-08-23 16:29:39.981772 [INFO] mod_dptools.c:3401 Originate Failed. Cause: INCOMPATIBLE_DESTINATION</div><div class="">freeswitch@iZ23lkvsnwpZ> </div><div class="">EXECUTE <a href="mailto:sofia/internal/1004@120.26.130.142" class="">sofia/internal/1004@120.26.130.142</a> answer()</div></div><div class="">-------------------------</div><div class="">It looks freeswitch send with ICE to webRTC. Does anyone know how i do disable the webRTC negotiation and only send normal sip info to webRTC gateway<span style="color: rgb(118, 118, 118); font-family: Arial, Helvetica; font-size: 13px; line-height: 20px;" class=""></span></div>_________________________________________________________________________<br class="">Professional FreeSWITCH Consulting Services: <br class=""><a href="mailto:consulting@freeswitch.org" class="">consulting@freeswitch.org</a><br class="">http://www.freeswitchsolutions.com<br class=""><br class="">Official FreeSWITCH Sites<br class="">http://www.freeswitch.org<br class="">http://confluence.freeswitch.org<br class="">http://www.cluecon.com<br class=""><br class="">FreeSWITCH-users mailing list<br class="">FreeSWITCH-users@lists.freeswitch.org<br class="">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br class="">UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br class="">http://www.freeswitch.org</div></blockquote></div><br class=""></div></body></html>