Loading page... (if page fails to load consider use Chrome)
 




dev

 
 
 
            


 
 
 
 

SIP/SIMPLE:
    WebRTC Clients on this server:
      group call


           ...or schedule one:
      :



      Group call participants:

       Leave group
       Send chat to everyone in group
      Autofocus Talker
       Mute yourself from group
       Mute everyone but yourself


      Call history:



       

      SDP BUNDLE  
      Template  
      REST auth use Template-2 
      Send DTMF as INFO 
       

       Send raw message:
       

       Simulate receive:
       


       Push notification token sender:
       
        

      User Account:
           (unsaved)

      Name | SIP addr. 
      Display name  
      SIP Auth. name  
      SIP Password  
      SIP Alias  
      Group(s)  
      List clients by groups 
      Show display names 
      Send presence status 
      Presence Status  
      Note  
       

       Media Sources:   
      Volume
      Ring signal   
      Ring phrase  
      Request HD camera 
      Stream HQ video 
      Cam always on 
      Classic look 

       
      Advanced settings

      Highlight errors 
      Open in tab 

       Conference server example link:
      SIP address  
      Pin code  

      Developer mode 


        



      Check your microphone!


      SIP registration failed!

      You should enter valid SIP credentials
      to be able to fully use the demo client.
      Please enter your name:

         
      Demonstration of the
      WebRTC & SIP PBX Companion
      View more...
        Alternative client:
        wrtcweb.ingate.se
      Demonstration of the
      WebRTC & SIP PBX Companion


        Ingate also brings these WebRTC features to the SIP PBX UC Infrastructure

        Clickable link to call me (pass via IM or e-mail):



        Clickable link to chat with me (with screen sharing on Chrome):


        Pass this link to join a conference:


        Add click-to-call button to your company website:
                Example page with Call Support button

                  Will call: (any SIP address or WebRTC name)

                           (exemplifies 800-number replacement)

                  ( to execute on second PC)

        Ease of creating new communication services:
        (which could be so much better with video)
                 Free Swedish 020 telephone calls   020 

        Powerful flexible client development kit SDK:
              Quick development of custom web clients.
              Integrated SIP support. From Javascript: wrtc.call("john.doe@company.com");

        HOW TO CONFIGURE AND TRY IT OUT                Read more at www2.ingate.com/inprogress/intro1


      Call
      Kick out
      Hang up
      Chat with
      Send file(s)
      Request screen
      Voice only
      Hold
      Group call
      Transfer call to
      Recall
      Mute Local
      Mute remote