The below WebRTC VoIP web client uses our online WebRTC-SIP gateway to convert the signaling and media between the browser WebRTC and your server SIP/RTP.
Just enter your SIP server address, SIP username and password to be able to register and make calls via your SIP server/PBX/Softswitch.

Notes: 
  • It will not work if your SIP server is behind NAT since this gateway is on the public internet and in this case it would not be able to connect to your server with private address. Download the free WebRTC-SIP gateway and install it on a machine near your SIP server if you wish to test with a SIP server located behind NAT.
  • If you don't have a SIP server, then you can test with our demo softswitch with the following settings:
    • server address: voip.mizu-voip.com
    • username: webphonetest2
    • password: webphonetest2
    • call to: testivr3

WebRTC-SIP gateway online demo