sipml

Unable to register to FreeSwitch server & unable to call SIP client (XLite) respectively using SIPml5 client

由 丶灬走出姿态 提交于 2019-12-25 04:53:11
问题 I am unable to register to FreeSwitch server & unable to call to SIP client (XLite) by using SIPml5 SIP client. Following is my HTML5 code: <!DOCTYPE html> <html> <head> <meta content="charset=utf-8"/> <script type="text/javascript" src="SIPml-api.js"></script> <title>SIP Client 1</title> <script type="text/javascript"> window.onload = function() { var readyCallback = function(e){ createSipStack(); // see next section }; var errorCallback = function(e){ console.error('Failed to initialize the

How to read Call-Info Header from Invite Message using sipml5

由 我只是一个虾纸丫 提交于 2019-12-13 07:19:36
问题 I use sipml5 with freeswitch and I need to detect when call should be answered automatically. The only part where I can get it from is SIP Invite message: recv=INVITE sip:username@IP:50598;transport=ws;intercom=true SIP/2.0 Via: SIP/2.0/WSS IP;branch=z9hG4bKd451.8dc49598935d4ebdf937de014cf1d922.0 From: "Device QuickCall"<sip:NUMBER@DOMAIN>;tag=68rtr6c12v9em To: <sip:michaltesar2@IP:50598;transport=ws> Contact: <sip:mod_sofia@IP:11000> Call-ID: dcd8fb4d69f0850840a743c152f4f7358a21-quickcall

SIPML 5 Client and SipServlets not works Using WSS

由 笑着哭i 提交于 2019-12-10 22:38:20
问题 I Have Tomcat run on HTTPS. I have tried to deploy SIPML5 WebSocket Application To into my tomcat. When I tried to connect Sip Servlets using ws : ws://192.168.X.Y:5082 And Sip Servlets Config looks like : <Connector port="5082" ipAddress = "192.168.X.Y" protocol="org.mobicents.servlet.sip.startup.SipProtocolHandler" signalingTransport="ws"/> I got an error : SIPml-api.js:4 Mixed Content: The page at 'https://192.168.X1.X2:8443/CallCenterBK/CallCenterBK.jsp?sip=1' was loaded over HTTPS, but

Changing a MediaStream of RTCPeerConnection

由 喜夏-厌秋 提交于 2019-11-28 18:23:49
I want to change from a audio/video stream to a "screensharing" stream: peerConnection.removeStream(streamA) // __o_j_sep... in Screenshots below peerConnection.addStream(streamB) // SSTREAM in Screenshots below streamA is a video/audio stream coming from my camera and microphone. streamB is the screencapture I get from my extension. They are both MediaStream objects that look like this: * 1 Remark But if I remove streamA from the peerConnection and addStream(streamB) like above nothing seems to happen. The following works as expected (the stream on both ends is removed and re-added)

Changing a MediaStream of RTCPeerConnection

由 末鹿安然 提交于 2019-11-27 11:18:46
问题 I want to change from a audio/video stream to a "screensharing" stream: peerConnection.removeStream(streamA) // __o_j_sep... in Screenshots below peerConnection.addStream(streamB) // SSTREAM in Screenshots below streamA is a video/audio stream coming from my camera and microphone. streamB is the screencapture I get from my extension. They are both MediaStream objects that look like this: * 1 Remark But if I remove streamA from the peerConnection and addStream(streamB) like above nothing seems
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