WebRTC (Client-less) Phone Test

To test if your Internet connection is OK and supports the WebRTC (Client-less, in-browser) SIP phone you can login to our test account and conduct 2 tests.

If any of the test fail please inform our support on which test failed.

  1. Login to the OCCP portal with any Chomium bases browser (Chrome, Firefox, MS Edge, etc -not Safari-)
  2. URL: https://satu.orencloud.com:8020/
  3. Username: 1000
  4. Password: webrtctest

After login you will see the WebRTC phone as shown in below screen-shots;-


Dial (star) *60, time announcement: this will prompt the system to play a time announcement, you should here simular as: "At the sound of the tone the time will be....."


Dial (star) *43, echo test: this will start a echo test meaning when you talk you will hear yourself back, this test will give 2 results, 1) if you hear yourself two way audio is working and 2) any voice delay, it is OK if there's a slight (less then 1sec) delay, however if the delay is more then 1 sec there might be a latency issue and we need to further investigate on the possible cause and solution.

Note: This test account can only call *43 and *60

Screenshot of the WebRTC phone



Article ID: 11
Last updated: 09 Feb, 2021
Revision: 5
Tools -> WebRTC (Client-less) Phone Test