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DoubangoTelecom / sipml5

The world's first HTML5 SIP client (WebRTC)
BSD 3-Clause "New" or "Revised" License
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getPlugin(…).createPeerConnection is not a function #333

Open E2-Dayanithi opened 4 years ago

E2-Dayanithi commented 4 years ago

I am using sipml5 webrtc client in my local to test webrtc. The extensions are registered successfully. But I am facing following error during call in all browsers.

TypeError: getPlugin(…).createPeerConnection is not a function at new RTCPeerConnection (VM306 adapter.js:471) at tmedia_session_jsep01.__get_lo (tmedia_session_jsep.js?svn=252:655) at tmedia_session_jsep01.tmedia_session.get_lo (tmedia_session.js?svn=252:583) at tmedia_session_mgr.get_lo (tmedia_session.js?svn=252:285) at tsip_dialog_invite.send_offer (tsip_dialog_invite.js?svn=252:288) at tsip_dialog_invite.send_invite (tsip_dialog_invite.js?svn=252:263) at tsk_fsm_entry.c0000_Started_2_Outgoing_X_oINVITE [as fn_execute] (tsip_dialog_invite__client.js?svn=252:90) at tsk_fsm.act (VM324 tsk_fsm.js:91) at tsip_dialog_invite.tsip_dialog.fsm_act (tsip_dialog.js?svn=252:750) at tsip_session_invite.call (tsip_api_invite.js?svn=252:155)

NikTNV commented 4 years ago

Have the same issue on RDP session.

rodrigocso commented 4 years ago

Had this issue myself, found the answer here: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=36804060 In short, it requires https to work. If you try to execute createPeerConnection from http, you'll get that error.

agenovez commented 4 years ago

As @rodrigocso said, you need https because it will not let you make the call without it (the button does nothing), you could try letsencrypt certificates