I have been working on setting up an HTML5 client (sipml5 by doubango: https://www.doubango.org/). The infrastructure of my setup is shown below:
- Server 1: sipml5 client, served through ngnix and https.
- Server 2: webrtc2sip setup with doubango, served over the secure tcp WebSocket, wss:\voip.example.com:10062
- Server 3: running FreeSWITCH setup with certificates, note: I am able to connect through a local sip client to my FreeSWITCH and make a call.
So far, I have been successful in getting the client connected to the web socket which in turn, connects to the FreeSWITCH. The issue arises when I try to make a call to another extension on the FreeSWITCH.
The abruptly fails with the following error message displaying on the web page:
Media stream permission denied
I am given the following error logs in the console of the browser:
State machine: c0000_Started_2_Outgoing_X_oINVITE
tsk_utils.js?svn=252:116 ICE servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.net:3478"},{"url":"stun:numb.viagenie.ca:3478"}]
tsk_utils.js?svn=252:116 ==stack event = m_permission_requested
tsk_utils.js?svn=252:116 ==session event = connecting
tsk_utils.js?svn=252:116 onGetUserMediaError
tsk_utils.js?svn=252:128 DOMException
tsk_utils_log_error @ tsk_utils.js?svn=252:128
tmedia_session_jsep01.onGetUserMediaError @ tmedia_session_jsep.js?svn=252:486
(anonymous) @ tmedia_session_jsep.js?svn=252:690
tsk_utils.js?svn=252:116 State machine: c0000_Outgoing_2_Cancelling_X_oCANCEL
tsk_utils.js?svn=252:116 PeerConnection::stop()
tsk_utils.js?svn=252:122 There is no INVITE request to cancel
tsk_utils_log_warn @ tsk_utils.js?svn=252:122
tsip_dialog_invite.send_cancel @ tsip_dialog_invite.js?svn=252:692
c0000_Outgoing_2_Cancelling_X_oCANCEL @ tsip_dialog_invite__client.js?svn=252:173
tsk_fsm.act @ tsk_fsm.js?svn=252:91
tsip_dialog.fsm_act @ tsip_dialog.js?svn=252:750
tsip_dialog.hangup @ tsip_dialog.js?svn=252:758
__tsip_dialog_invite_media_callback @ tsip_dialog_invite.js?svn=252:894
tmedia_session_mgr.callback @ tmedia_session.js?svn=252:109
tmedia_session_jsep01.onGetUserMediaError @ tmedia_session_jsep.js?svn=252:487
(anonymous) @ tmedia_session_jsep.js?svn=252:690
tsk_utils.js?svn=252:116 ==stack event = m_permission_refused
tsk_utils.js?svn=252:116 ==session event = terminating
I am not getting any error logs from either the FreeSWITCH server or the webrtc2sip server. Both were throwing errors when I was debugging the initial FreeSWITCH registration between sipml5 -> webrtc2sip -> FreeSWITCH.
I should also mention that I have enabled microphone access for the website in chrome. I will also mention that these are all internal servers only accessible over our office LAN.
UPDATE: I should also mention that I am registered successfully to the freeSWITCH server:
freeswitch@voip.example.local> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
1000,voip.example.local,614eb848-c522-1514-2fb6-1d4fa5ad4f1e,sofia/internal/sip:1000@10x.x.xx:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=10x.x.xx;ws-src-port=2049;ws-src-proto=wss,1548964623,10x.x.xx,10060,udp,voip.example.local,
Hoping to find a fix or at least a workaround for now. Any help appreciated.
