Wiki -VoIP Topics

WebRTC-SIP Gateway Demo


 
The below linked 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.
This demo uses the mizu webphone WebRTC client, howerver you are free to use the gateway with any WebRTC client such as sipml5, sipjs, jssip and others.

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
    • (or connect with another SIP or WebRTC client using webphonetest1/webphonetest1 and make calls between them)
  • In case if you are interested in a gateway demo (not just see a demonstration running our WebRTC client which works via our test gateway) then you can just download the free version from here and install it on your server to test it.

 

 

Proceed to WebRTC client demo page

(use the Softphone page which works via the demo WebRTC gateway)